Haagen, J.

Hansen, H. B.

Larsen, Knud J.

Mikkelsen, K. B.

Nielsen, H.

Online Research Database In Technology

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B.; Nielsen, H.; Larsen, Knud J.Published in:Proceedings of the IEEE Workshop on Speech Coding for TelecommunicationsPublication date:1995Document VersionPublisher's PDF, also known as Version of recordLink back to DTU OrbitCitation (APA):Hansen, H. B., Haagen, J., Mikkelsen, K. B., Nielsen, H., & Larsen, K. J. (1995). Toll Quality Codec for the GSM-System. In Proceedings of the IEEE Workshop on Speech Coding for Telecommunications (pp. 59-60). IEEE.Toll Quality Codec for the GSM-System Hen& B. Eansen, Jesper Haagen, Keld B. Mikkeben, Henrik Nielsen, and Knud J. Larsen’. Tele Danmark Research, Lyngs0 AU6 2, DK-2970 H0rsholm, Denmark. Institute of Telecommunication, Technical University of Denmark, DK-2800 Lyngby, Denmark. ABSTRACT This paper describes the basic performance requirements for a new enhanced GSM codec and a proposal for this appli- cation based on the ITU 8 kb/s coder standard. A powerful channel wding scheme using more than half the gross b i t  rate is designed and the performance of the codec ir formally evaluatedusing MOS testing. The overall system is found to provide toll qudity speech with only a graceful degradation at increasing bi t  error rates, and it meets the preliminary quality requirements for the new standard. 1 Introduction In Europe, a new codec for the GSM Wireless communica- tion system is required. Currently two codecs have been standardized: the GSM full-rate codec [l] at 22.8 kb/s gross bit rate, and the GSM half-rate codec at 11.4 kb/s [2]. Since the introduction of the GSM full-rate codec, the advances in speech coding and VLSI technology have made high quality coding for the full-rate channel feasible. The high quality codec is targeted mainly at new markets like the evolving PCS (Personal Communication Systems), but it i s  envisaged that an enhanced fnll-rate codec could be included as an option in multi-mode GSM terminals sup porting both full-rate and half-rate coding schemes. In any case, the growing demand for mobile comm.unication stresses the need for speech coders capable of providing toll quality speech on wireless connections at least at relatively low error rates. Since a relatively tight time schedule is foreseen for the development of the high quality codec (less than one year), it would be convenient if an existing codec could be adapted to function in this wireless environment. In general, high quality codecs are more sensitive to channel errors than the existing GSM full-rate (FR) codec. This means that a new enhanced speech codec should offer better quality at somewhat lower bit rate allowing additional bits to be allocated to channel coding. In this paper we describe a channel coding scheme which can be used in conjunction with the new toll-quality ITU 8 kb/s codec [3] and the experiments carried out to test the robustness of the codec combination in the GSM environ- ment. The following issues will be address& 0 GSM enhanced full-rate (EFR) coding performance re- 0 channel coder design 0 error detection and concealment techniques algorithmic delay quirements 2 Performance requirements The essential requirements [4] for the high quality codec indude speech quality, compIexity, and delay. The speech quality should be comparable to that of G.728 [SI for error free conditions and conditions with relatively few channel errors. Under severe error conditions the quality should be significantly better than that of the GSM full-rate codec. Concerning the complexity of the codec it should not ex- ceed significantly that of the GSM half-rate codec, i.e. 4 times the full-rate codec complexity. The coding delay, including time-slot interleaving, is lim- ited to that of the GSM full-rate system, i.e. 90 ms end- to-end delay. However, in order to mitigate the echo can- cellation problems it is a deskable requirement to limit the coding delay to 40 ms. 3 Channelcoding The channel coding scheme is very similar to those already implemented in the GSM system. It is basically a low con- straint length convolutional coding scheme without tail bits transmitted. The low speech coder bit rate allows a channel coding rate of almost 113 for all bits in each 10 ms frame. The speech coder bits are divided into two classes contain- ing 54 bits and 25 bits, respectively, excluding the parity bit which is treated differently as explained below. The two classes are encoded at rates R = 6/17 and R = 113, respec- tively. Even though the tail bits are not transmitted, a low constraint length CL = 4 corresponding to 8 coding states is chosen. Preliminary experiments have shown that perfor- mance does not really benefit from increasing the constraint length for the convolutional code. A low constraint length code recovers faster from the occasional overloading during channel fades, and it also ensures a fairly low channel coda compl&ty. 3.1 Error detection Error detection is based on soft decision information. Both input channel soft decision and channel decoder output soft decision information is used to produce the bad frame in- dication. The parity check included in the speech coder 59 Authorized licensed use limited to: Danmarks Tekniske Informationscenter. Downloaded on March 02,2010 at 04:57:31 EST from IEEE Xplore.  Restrictions apply. itself is not very useful in this context because single errors rarely occur during fades. Therefore the parity bit is used for channel coding and after the channel decoding parity is set to either check-or fail depending on the channel decoder output. This technique actually adds some flexibsty to the error concealment because in case of a parity error only the pitch parameters are replaced in the decoder. This means that a parity error may be utilised to signal a slightly cor- rupted frame which should not be totally discarded. The speech decoder also recognises bad frames flagged by the error detector. These frames are discarded and the speech decoder extrapolates the output signal. candidate EFR208 EFRlO6 EFRlO4 4 Delay frame size depth delaJr 20ms 8 FR + 5 ms 10 ms 6 FB - 35 ms 10 ms 4 FR - 55 ms The algorithmic delay depends on the speech coder fkame size and the channel burst interleaving depth. If two speech coder frames are merged into one 20 ms frame and then interleaved over eight bursts, the configuration is similar to that of the FR system. However, due to the speech coder sample look-ahead the delay limit will be exceeded slightly. A reduction in the total delay can be o b t k e d  by reducing the i n t e r l e a ~ g  depth. For a frame size of 10 ms and inter- leaving depths less than eight the algorithmic delay is below the limit. Some of the possible configurations of frame size I I interleaving I Table 1. Algorithmic round-trip delays and interleaving depth are shown in Table 4 and these can- didates are covered by the test desaibed below. 5 Subjective test The speech coder itself is already characterised during the standardisation process. The purpose of this MOS (Mean Opinion Score) test is to assess the quality of the coding system under bad channel conditions. In order to  measure the effect of a delay reduction, three H e r e n t  interleaving depths are included in the experiment as separate condi- tions. The G728 and GSM FR standard coders are induded as references. The G728 is tested in error-free condition only while both the EFR candidates and the GSM FR are s u b  jected to three different error patterns in addition to the error-free condition. Figure 1 shows the subjective test results. The dotted line represents the reference quality level corresponding to the measured G728 quality on dear channel. The listeners have consistently given rather low scores throughont the ex- periment. The reason for this could be that the experiment is not balanced sufficiently well because a major part of the speech samples is almost equal in quality. 6 Conclusion In this paper the application of the ITU 8 kb/s speech coder in a GSM EFR coding system is addressed. A channel coder 4 1  I EFR EFR 2 EFR 1 GSM 104 ,.m fnf. 1WB 7dB 460 CA Figure 1. Subjective test  results. to accompany the speech coder is described and the perfor- mance of the coding system in bad channel conditions is in- vestigated by means of subjective testing. The test results show that by appl*g a powerful channel coding scheme it is possible to  retain a quality level dose to toll quality for the bad channel situations. Besides speech quality, the amount of delay introduced by the codec is an important factor when the goal is to achieve performance approaching that of a wireline network. A s  the speech coder has a fairly low algorithmic delay of 15 ms, a significant reduction in total delay could be achieved if the burst interleaving is removed. However, since the error cor- rection benefits very much from the burst interleaving, this requires very strong channel coding if speech quality is to be retained. Other aspects like background noise performance and overall complexity need to be investigated further be- fore a find conclusion on applicability to  GSM EFR can be reached. So far the results suggest that a solution offering the requested quality with some reduction in dehy can be found. References [l] ETSI, “GSM Recommendation 06.1D: Full-Rate Speech 121 ETSI, “GSM Recommendation 06.20: Speech Coding for [3] ITU-TSS Study Group 15, Draft Recommendation G.729, [4] SMG1, “EnhanccdFull-Rate Spccch Codec Study Phase Re- port,’’ ETSI SMG-TC, Dec 1994. [SI ITU-T, Recommendation G.728. [6] H. B. Hansen, K. 3. Larsen, H. N i h ,  andK.  B. Mikkelsen, “Channel Coding Schemes for the GSM Half-Rate System,” in Speech and Audio Coding for Winlcss and Network Ap- plications, edited by Bishnu S. Atsl, Vladimir Cupuman, and Aucn Gersho, pp. 171-179, Kluwer Academic Publish- M, 1993. [7] K. B. Mikkelsen, K. J. Larsen, H. B. Hansen, H. Nielsen, and H. Jacobs-, ”The Danish Candidate for the GSM Half- Rate Speech Channel,” SIGNAL PROCESSING VII, Pmc. of EUSIPCO-9.& pp. 995-998, EUMSIP 1994. Transcnding,” Sep 1988. The GSM HaIf -h te  Channel,” &E&, Jan 1995. June 1995. 60 Authorized licensed use limited to: Danmarks Tekniske Informationscenter. Downloaded on March 02,2010 at 04:57:31 EST from IEEE Xplore.  Restrictions apply. 

Toll Quality Codec for the GSM-System

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Toll Quality Codec for the GSM-System

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