Improvement of tandemless transcoding from AMR to EVRC

Transcription

1 Iproveent of tandeless transcoding fro to Wonil Lee, Sunil Lee, Changdong Yoo Dept. of Electrical Engineering and Coputer Science, KAIST ABSTRACT The siplest ay to transcode to different speech coding standards is by the tande ethod, but this results in quality loss, coputational increase and etra delay. This paper proposes a tandeless transcoding ethod for Adaptive Multi Rate ()[] hich is the speech coding standard of the asynchronous IMT-000 syste and Enhanced Variable Rate Codec ()[]. The proposed algorith is coposed of three parts : () conversion using linear interpolation, () pitch conversion using ecitation signal and (3) adaptive / fied codebook gains conversion. The proposed algorith reduces not only processing delay but introduces less degradation to speech than that of the tande ethod, hich as verified by an objective easure.. INTRODUCTION Generally, the siplest ay to solve the copatibility proble beteen the to different speech coding standards is by the tande ethod : generate speech signal using a decoder of one standard and then re-encode the signal by the other standard. This approach results in quality loss, coputational increase and etra delay since the speech signal undergoes encoding and decoding processes tice. These probles can be solved by the tandeless transcoding ethod in hich the paraeters encoded by one standard is directly decoded by the decoder of the other standard. This paper proposes a tandeless transcoding ethod for and. The proposed ethod directly converts the paraeters that are coon in both coders. The proposed algorith consists of three parts : () conversion using linear interpolation, () pitch conversion using ecitation signal and (3) adaptive / fied codebook gains conversion. The speech signal of better or at least sae quality can be obtained ith the reduced coputation and delay if the tandeless transcoding ethod is used. The rest of the paper is organized as follos. In section, the siilarities and differences of and are eained. Section 3 of the paper discusses the ays to convert the paraeters encoded by to those of. Section 4 provides the results of objective evaluation on the proposed ethod. Section 5 rap up the paper and briefly describes ongoing ork.. VERSUS is based on Algebraic CELP (ACELP), and it is the speech coding standard of the asynchronous IMT-000 syste. operates in eight different bit-rate odes ranging fro 4.75 kbit/s to.kbit/s. On the other hand, based on Relaed CELP (RCELP)[5] operates in three different bit-rate odes ranging fro 0.8 kbit/s to 8.5 kbit/s. The siilarities and the differences of and are suarized in Table.

2 Algorith Analysis size LPC analysis quantization Pitch Fied codebook Gains Analysis by Synthesis 0s frae size 4 subfraes (5s) 5s or no lookahead Predictive Split VQ (,,,,)/(3,3,4) Update every subfrae Fractional pitch ACELP Scalar quantization Vector quantization Generalized Analysis by Synthesis (RCELP) 0s frae size 3 subfraes 0s look-ahead Split VQ (,,3,3)/(3,3,4) Update every frae Integer pitch ACELP Scalar quantization Table. Siilarities and differences of and As indicated in Table, the ajor differences of the to standards are in the length of subfrae, the pitch search and the quantization of LPC coefficients and the gains. 3. PROPOSED ALGORITHM The tandeless transcoding ethod fro to consists of the consists of three parts : () conversion using linear interpolation, () pitch conversion using ecitation signal and (3) adaptive / fied codebook gains conversion. 3. CONVERSION OF and operate on speech fraes hich have the sae length, 0s, hoever the length and the shape of the indos for linear prediction analysis and the length of lookahead are different. Figure describes the conversion process using linear interpolation. First, the coefficients of the th frae is obtained fro the bit-strea encoded by Coder (either or ) using its decoder. Second, the decoded coefficients of the th frae and those of the (-) th frae is linearly interpolated in order to obtain the coefficients of the current frae of Coder (either or ) as represented by Ω ( ) B ( µ ) Ω + µ Ω () ( ) A ( ) A here, and are the 0-th order coefficients of the th frae decoded by Coder and Coder, respectively. The eighting constant µ ranges fro 0.8 to 0.9. bitstrea Decoder () Encoder () ( ) Ω A Interpolation Interpolation ( ) Ω B Encoder (EV RC) Decoder (EV RC) Figure. Conversion of beteen and EV RC bitstrea When linear interpolation of is perfored, the coputation cost is reduced in coparison to that of the tande ethod since the pre-processing, the linear prediction analysis and the quantization of the coefficients that are required in the decoding/encoding processes of cross-tandeing ethod are not necessary. There is no additional quantization error since the coefficients are converted directly. 3. CONVERSION OF PITCH is based on ACELP and it finds the integer pitch delay through open-loop pitch search. It then attains the fractional pitch delay by searching near the integer pitch value ith the resolution of /6 or /3. perfors the pitch search for every other subfrae. In coparison, based on RCELP finds only one integer pitch delay for every other frae. The pitch of the subfrae is calculated by linear interpolating the integer pitch value of the frae. Then these pitches are linearly interpolated to calculate the delay contour. allocates ore bits to the ecitation signal than the pitch paraeter to prevent the degradation of quality of the speech signal. The key point of the conversion of the pitch is to convert the four pitch values obtained fro one frae of to one integer pitch value hich requires. A

3 siple solution is to interpolate the four pitch values of linearly to obtain the pitch of. But this approach results in the serious degradation in speech quality. We propose the conversion perfored in the ecitation doain. First, the adaptive codebook and fied codebook signals are decoded fro the bit-strea encoded by Coder and added to calculate the ecitation signal. The pitch delay of is calculated fro this ecitation signal. This process is depicted in Figure. inde bitspara ( ) Pitch inde Gains inde FCB inde ACB Decoder Gain Decoder FCB Decoder v[n] c[n] ĝ p ĝ c u[n] u[n] (-) open-loop pitch sench target signal coputation ACB gain Coputation Figure. Conversion of pitch and adaptive codebook gain Originally, the pitch of is calculated fro the residual signal obtained by inverse filtering the preprocessed speech signal. We consider the ecitation signal fro as this residual signal and find the integer pitch delay for. This approach significantly iproves speech quality copared to the direct conversion. The coputation cost can be reduced by this approach because not the speech signal but the only ecitation signal is decoded. 3.3 CONVERSION OF ADAPTIVE AND FIXED CODEBOOK GAINS Although the notations are different, the adaptive codebook gain( g ) and the fied codebook gain( g ) of and have the sae eaning. Both the gains of and are defined as the cross correlation beteen the target signal(defined later) and the signal that is coputed by filtering the signals of the adaptive or fied codebook through the eighting synthesis filter. The adaptive and fied codebook gains of and are given by p c pitch gain p t Hv t t v H Hc t Hc t t c H Hc g () g (3) c here v and c are adaptive and fied codebook vectors, respectively. The eighting synthesis filter is given by H A( z γ) H ( z) (4) A( z) A( z γ ) here γ, γ have the value beteen 0 and, 0.9 and 0.6 in, 0.9 and 0.5 in, respectively. The target signal for the adaptive codebook search hich is coputed by subtracting the zero input response of the eighted synthesis filter fro the eighted speech signal. The second target signal for the fied codebook search is coputed by subtracting the adaptive codebook contribution fro. The to target signals and here are given by H zir s, g Hv (5) H zir synthesis filter. p is the zero input response of the eighting In, the gains are scalar quantized in. kbit/s and 7.95 kbit/s odes and vector quantized in the other ode. The 4 th order MA(Moving Average) prediction is used for the quantization of the fied codebook gain. In contrast, all the gains of are vector quantized. Figure 3. Fied codebook gains of and

4 The conversion of the fied codebook gain is siple. The fied codebook gain hich is decoded fro the bitstrea encoded by is nearly sae as the fied codebook gain of if it is scale-upped (see Figure 3). But the direct conversion of the adaptive codebook gain degrades the quality of the speech signal seriously. So e obtain the adaptive codebook gain using the ecitation signal like the case of the conversion of the pitch. This approach doesn t result in the quality loss. 4. PERFORMANCE EVALUATION As an objective easure, e use the Average Spectral Distortion Measure(ASDM) that easures the root ean square difference fored beteen the to log-poer LPC spectrus. ASDM is given by ASDM π here, π 0 M [0 log (6) 0 M S ( ω ) 0 log 0 Sˆ ( ω )] dω is the poer LPC spectra of the -th LPC analysis frae hich is directly encoded by, and S (ω) ˆ (ω) S represent the poer LPC spectra of the -th LPC analysis frae hich is transcoded fro using the cross tandeing ethod or the tandeless ethod. M denotes the total nuber of LPC fraes used to calculate ASDM. In practice, S (ω) and S ˆ ( ω) are calculated ith a finite resolution provided by a N-points FFT. Speech data consists of the to sae sentences spoken by ale and feale, here both are of about 3 second long. We transcode the speech signals fro to using the cross-tandeing ethod and the tandeless transcodidng ethod respectively to calculate the ASDM value. Although there are any odes in both and, e use 7.95 kbit/s ode of and Rate of because the bit-rate of these odes are siilar kbit/s ode of and Rate / of are also evaluated. Table shos the result of ASDM calculation. 7.95kbps 4.75kbps Rate/ ASDM Rate (db) Feale Male Feale Male Tande Tandeless Table. ASDM results of for the cross tandeing and the tandeless ethod As shon in Table, the ASDM values of the tandeless transcoding ethod are less than those of the cross tandeing ethod for both sentences spoken by ale and feale. That is, the LPC spectru of the speech signal transcoded by the tandeless ethod is ore siilar to that of the speech signal hich is directly encoded by than that of the speech signal transcoded by the cross tandeing ethod. 5. CONCLUSION We proposed the tandeless transcoding ethod fro to. The tandeless transcoding ethod consists of the conversion of using the linear interpolation, the conversion of the pitch for the adaptive codebook and the conversion of the gains. We could reduce the coputation cost and the delay ithout any degradation in speech quality ith the proposed ethod. The ethod to convert the fied codebook should be eained as further study. ACKNOWLEDGEMENT This research as supported by Korea Science and Engineering Foundation (KOSEF) under Grant for hich the authors ould like to epress their thanks. REFERENCES. 3GPP TS 6.090, Speech Codec; Transcoding functions Dec. 999

5 . TIA/EIA/IS-7, Enhanced variable rate codec, speech service option 3 for ideband spread spectru digital systes, Hong-Goo Kang,; Hong-Kook Ki, R.V.Co, Iproving transcoding capability of speech coders in clean and frae erasured channel environents, IEEE Proc. 000 IEEE Workshop on Speech Coding, 000., pp , Jan., S.W. Yoon, An Efficient Transcoding Algorith For G.73. and G.79A Speech Coders, M.S. dissertation, Yonsei University, W.B.Kleijn, P.Kroon, The RCELP Speech-Coding Algorith, European Trans. on Teleco., vol. 5, no 5, pp , Sep./Oct., 994