An Efficient Variable-Bit-Rate Low-Delay CELP (VBR-LD-CELP) Coder

Digital circuit multiplication equipment (DCME) permits the concentration of a number of 64kbit/s PCM encoded input trunk channels on a reduced number of transmission channels. It has two basic techniques; one is digital speech interpolation (DSI), the other is adaptive differential pulse code modulation (ADPCM). Essential facilities, interface conditions and overall performance requirements are given by recommendation G.763, 1988. However, frame structure didn't detail assignment. Many corporations such as ECI, NEC and AT and T have produced products. They have used different DCME frame structures. NEC'S NCM-501 DCME fully complies with INTELSAT IESS-501, DCME multiframe length is is longer than new DCME frame length. This new DCME frame structure is described. >

An advanced adaptive differential pulse-code modulation (ADPCM) technique applicable to digital circuit multiplication equipment (DCME) is described. The performance of the advanced ADPCM CMOS single-chip codec regarding voice and various high-speed modems is discussed. Major issues in DCME are also addressed. The speech quality of the 32 kb/s advanced ADPCM is equivalent to that of CCITT G.721 ADPCM and also provides better quality at 24 kb/s. The 32 kb/s advanced ADPCM can transmit V.29 modem signals at 9600 b/s over two asynchronous tandem links and V.32 modem signals over four asynchronous tandem links, respectively. Evaluation results indicate that at 40 kb/s, the advanced ADPCM again provides superior performance to the 40 kb/s G. 721-based ADPCM for V.33 modem at 14.4 kb/s. It is noted that an advanced ADPCM codec having a variable rate coding capability between 24 and 32 kb/s effectively can be applied to DCME allowing a higher gain and better speech quality to be achieved. >

A highly efficient algorithm termed adaptive forward-backward vector quantization (AFBVQ) is developed for variable bit rate quantization of linear predictive coding (LPC) coefficients and integrated with the FS1016 Federal Standard Code Excited Linear Predictive (CELP) coder. This results in a high performance low bit rate speech coder called as AFBVQ-CELP which brings in two-fold bit rate reduction by backward LPC indexing and by forward LPC VQ. In AFBVQ, a previously decoded and temporally close speech signal is re-segmented into overlapping blocks. As the LPC coefficients calculated from one of those synthetic blocks are spectrally close to the current unquantized LPC coefficients, the backward LPC indexing is used to encode the current speech block; otherwise, the forward linear prediction is practised with the split vector quantization supported by a very efficient codebook initialization termed Mixture Gaussian Clustering (MGC). When compared to FS1016 CELP coder, AFBVQ-CELP reduces the LPC bit rate by 18 bit-per-frame (bpf) at the same spectral distortion. It means the overall bit rate is reduced from 4.8 kbps (FS1016 CELP) to 4.2 kbps. Furthermore, the proposed AFBVQ consistently outperforms the traditional forward LPC VQ by 3 bpf with the same spectral distortion. Subjective listening tests show that with AFBVQ-CELP the LPC bit rate can be further reduced to 8.4 bpf, resulting in 3.94 kbps overall bit rate without compromising the decoded speech quality.

Optimum quantization and code assignment schemes are described which minimize the subjective quality degradations introduced into the output speech of code excited linear predictive (CELP) coders by channel degradations. The background and basis for use of minimum redundancy for error control are examined. Greater emphasis is placed on adjustment of corrupted parameters to minimize subjective degradation rather than outright bit-by-bit error correction. These schemes are mostly tested on the CELP baseband coder, but they should be applicable to any linear predictive coder. They raise the bit rate of a 4.8-kb/s coder by about 12.5% and its MOS at 2*10/sup -2/ bit error rate by about 21.1% (scale 1-5). >

An error protection scheme for a vector-quantization (VQ)-based code excited linear predictive (CELP) coder for mobile radio applications is proposed. To arrive at an appropriate error protection strategy, the sensitivity of the different parameters in the CELP coder are studied, and then unequal error protection is designed according to parameter sensitivities. Different levels of error protection are provided by the use of punctured convolutional codes. The mobile radio channel is modeled as a Rayleigh fading channel. A convolutional interleaving technique is used to combat fading. The combined source and channel codec for this application has a total bit rate of 6.4 kb/s, of which 1.6 kb/s is used for channel coding. Simulation result show that the resulting codec can provide good quality speech at a Doppler frequency of 48 Hz and a channel bit error rate as high as 10/sup -2/. >

The problem of tandem vocoders and their effect on voice quality is well-known. In the case of international mobile-to-mobile calls, there can be up to three vocoding stages - between each mobile and its base station controller, as well as a digital circuit multiplication equipment (DCME) or voice over IP (VoIP) stages on the international link. The ITU standards for 2G and 3G networks define the concept of tandem free operation (TFO) for a mobile-mobile call. The same concept can be applied to the three-vocoder-stage international call, if the DCME or VoIP devices are capable of simulating TFO-capable mobile equipment. In the case of an international call from a mobile to a landline, a two-stage tandem of vocoder can be replaced with one pair of vocoders if the DCME or VoIP device pretends to be a TFO-capable mobile network.

The CCITT expert group on speech quality, formed in part to derive methodologies for evaluating new speech technologies, is emphasizing the assessment of digital circuit multiplication and packetized voice systems. Study group XII was requested to assess the performance of digital circuit multiplication equipment (DCME). The authors discuss DCME terminology, applications, testing alternatives, and issues that need to be addressed. >

This paper illustrates the salient features and the technical structure of the digital circuit multiplication equipment (DCME) specified by INTELSAT. First, an overview of the basic DCME requirements is presented and the network applications of the DCME are discussed. Then a systematic and detailed description of the body of the technical specification is provided. Finally, the complementary activities that followed the completion of the specification are briefly addressed.

Summary form only given. A compression algorithm for high quality speech signal using predictive coding techniques is developed. Code-excited linear predictive coding (CELPC) is one of the key techniques to compress speech signal to a bit-rate around 4.8 Kbps. However, due to the heavy computational requirement in the CELPC and speech signals usually can be divided into two portions: namely the based-band and the high-band frequency range. A hybrid CELPC and voice excited linear predictive coding (VELPC) scheme is developed for speech coding to reduce the complexity of the original CELPC. In the algorithm, a speech signal is firstly divided into two portions, the based-band and high-band respectively, in frequency domain, and then the low portion is coded with CELPC and the high-band portion is coded with VELPC. The test experiments showed this new coder can produce synthesized speech with good quality at a better bit rates than the original CELPC. When using the coding methods for the base-band and the high-band signal, we must decide how to divide the speech signal into two portions. In choosing the bandwidth of the base-band signal, there is a trade-off between the coding quality and the bit rate. In our experiment, the bandwidth of the base-band signal is chosen as one fourth of that of the original speech. Subjective evaluation experiments were conducted to test the performance of the hybrid CELPC and VELPC technique. For speech signal sampled at 8 kHz, a bit rate of 4.0 kbps can be achieved with frame intervals of 23 ms. The experimental results showed that the quality of the synthesized speech using hybrid coding technique at the bit rate of 4.0 kbps was almost the same as that of the CELPC at the bit rate of 4.8 kbps.

The code-excited linear predictive (CELP) coder is considered to be a good candidate for speech coding at low bit rates. Although the speech quality of this coder is good, certain distortions can still be perceived. In the paper, the authors introduce some procedures to improve the performance of the CELP coder. They improve synthetic speech quality quite a lot both in objective (signal-to-noise ratio) and subjective (listening tests) measures. Experimental results revealed that the proposed CELP coder improved segmental SNR by nearly 3 dB over the conventional CELP coder at 5.2 kbit/s. >

This paper describes applications of adaptive predictive coding (APC) with maximum likelihood quantization (MLQ) which can cover a wide range of coding rates from 4.8 to 16 kb/s for low C/N satellite communication systems, such as maritime, aeronautical mobile and thin‐route satellite communication systems, and also for speech and data integration, including digital circuit multiplication equipment (DCME) in business communication systems, such as INTELSAT business services (IBS). A 16 kb/s APC–MLQ hardware codec has been implemented by NEC–7720 DSP chips and the performance has been confirmed in subjective quality of speech through conversational tests. The objective performance has also been evaluated for non‐voice signals, such as single and multi‐frequency tones, and 1200 and 2400 b/s voiceband data signals. The APC‐MLQ codec can transmit the voice‐band data at 1200 b/s over two asynchronous tandem links and at 2400 b/s over one link. It was noted that the APC‐MLQ codec is superior in speech performance at 16 kb/s to a narrow‐band companded FM and meets requirements for low C/N satellite communication systems. For voice and data integration into 16 kb/s for 64 kb/s links, we propose a multi‐media multiplexing for low C/N digital satellite communication systems and also a small‐scale circuit multiplication system for business use. In these systems, a variable rate coding of APC‐MLQ from 4.8 to 16 kb/s can be effectively introduced for voice and data integration.

The code-excited linear predictive (CELP) coder is considered to be a good candidate for speech coding at low bit rates. Although the speech quality of this coder is good, certain distortions can still be perceived. The long-term predictor is a crucial part in CELP coder. The authors introduce some procedures to find the long-term predictor parameters. They improve the synthetic speech quality quite a lot both in objective (signal-to-noise ratio) and subjective (listening tests) measures. >

The LD-CELP (code excited linear prediction) algorithm was adopted by the CCITT, as a Recommendation G.728 for the coding of speech with toll quality at 16 kbit/s. The operation of the LD-CELP algorithm at 12.8 kbit/s is described, and its performance is assessed both with voice and nonvoice signals in single and interconnected network configurations. The 12.8 kbit/s LD-CELP codec is found to perform equivalently to 24 kbit/s ADPCM (adaptive differential pulse code modulation). It is also shown that 12.8 kbit/s LD-CELP is acceptably transparent to network signaling. As a result, it can be concluded that the operation of the LD-CELP algorithm at 12.8 kbit/s presents a viable option for inclusion within 16-kbit/s-based DCME (digital circuit multiplication equipment) or PCME (packet circuit multiplication equipment) overload strategies.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Recent years have seen widespread deployment of packetized circuit multiplication equipment (PCME) and digital circuit multiplication equipment (DCME) on international transmission facilities. Both use digital speech interpolation and low rate encoding to increase traffic concentrations on the available facilities and are designed to achieve compression ratios of 5:1 or more for voice traffic. If a significant amount of facsimile traffic is being carried, however, the advantages are significantly reduced. The Facsimile Demodulation and Compression Protocol (FADCOMP) specified as part of CCITT Recommendation G.765 and its implementation and performance in the AT&T integrated Access and Crossconnect System (IACS) are described. Field data are presented to demonstrate compression ratios similar to those for voice, while service transparency is maintained. >

A 4.8-kb/s delayed decision code excited linear prediction (CELP) coder that uses tree coding is described. In conventional CELP coding, short-term and long-term prediction parameters as well as excitation parameters are sequentially determined. In the proposed delayed decision CELP coding, a tree coding method is utilized. The long-term prediction and excitation parameter candidates obtained in each subframe are listed as a tree and the optimum combined parameter sequences are selected to minimize global quantization distortion over the coding frame. The proposed coding method significantly increases the quality of the 4.8-kb/s CELP coder at the cost of an additional 5-ms coding delay. >