Showing papers on "Enhanced Variable Rate Codec published in 1994"
TL;DR: A toll quality speech codec at 8 kb/s suitable for the future personal communications system and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements.
Abstract: A toll quality speech codec at 8 kb/s suitable for the future personal communications system is presented. The codec is currently under standardization by the ITU-T (successor of CCITT) where the codec terms of reference were mainly determined considering PCS application. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP) and has a speech frame of 10 ms. Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Formal subjective listening tests, performed by ITU-T SG 12, showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kb/s in error-free conditions and it outperforms G.726 under error conditions. The codec performs adequately under tandeming conditions, and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements, and this is one subject of study for the next phase. The algorithm has been implemented on a single fixed-point DSP for the ITU-T subjective rest, and required about 29 MIPS. An optimized version, however, requires 24 MIPS without any speech quality degradation. >
110 citations
19 Apr 1994
TL;DR: The ANT approach for the standardisation of the GSM half rate codec is presented and the use of error concealment at parameter level in the channel decoder as well as at signallevel in the speech decoder is discussed.
Abstract: The ANT approach for the standardisation of the GSM half rate codec is presented. The speech codec uses efficient scalar LSP quantization, joint optimization of adaptive and fixed codebook signals and works in two modes using different bit rates (5.7 and 6.15 kbit/s). The advantage of this dual mode scheme is an increase of the average error robustness without degradation of the intrinsic speech quality. The channel codec is based on rate compatible punctured codes. The channel decoder generates soft information for each decoded information bit. Bad frame detection is exclusively based on the exploitation of soft decision information. The use of error concealment at parameter level in the channel decoder as well as at signal level in the speech decoder is discussed. The complexity of speech and channel codec is 3.8 times the GSM full rate codec complexity. Subjective tests showed that the average Q-value of all test conditions is 1.7 db below the average GSM full rate Q-value. >
11 citations
NEC1
TL;DR: A channel coding method for a half-rate speech codec that includes new error control techniques that effectively integrates a speech codec with a channel codec and employs temporal correlation in encoded speech data as well as the redundancy of convolutional code, to improve its decoding performance.
Abstract: A channel coding method for a half-rate speech codec is proposed. This method includes new error control techniques that effectively integrates a speech codec with a channel codec. It utilizes bit swelling technique for an efficient unequal error protection. Highly significant bits are swollen and combined with other significant bits. Combined bits are coded with a convolutional code. In addition, the Viterbi decoder employs temporal correlation in encoded speech data as well ail the redundancy of convolutional code, to improve its decoding performance. The paper shows the error correction ability for this method is higher than that for a conventional method. A subjective test shows that the speech quality for a half-rate speech codec combined with the proposed channel codec is considerably better than that for the full-rate VSELP codec on a fading channel. >
7 citations
01 May 1994
TL;DR: This paper describes a full-custom ASIC implementation of the QCELP voice compression algorithm, which contains a custom DSP core, utilizing internal RAM and ROM and a co-processor, which were optimized to perform QCelP and related voice-band processing.
Abstract: This paper describes a full-custom ASIC implementation of the QCELP voice compression algorithm. Applications for the ASIC are code division multiple access (CDMA) based cellular, personal-communications-networks (PCN), and wireless local loop. The ASIC contains a custom DSP core, utilizing internal RAM and ROM and a co-processor, which were optimized to perform QCELP and related voice-band processing. Key design concerns were power consumption, area, flexibility, and testability. >
4 citations
19 Apr 1994
TL;DR: The design of a speech and channel codec for the North American TDMA digital cellular half rate channel which meets the eligibility requirements of the Telecommunications Industry Association (TIA) contest is described.
Abstract: The design of a speech and channel codec for the North American TDMA digital cellular half rate channel which meets the eligibility requirements of the Telecommunications Industry Association (TIA) contest is described. Its design objectives are largely shaped by the selection criteria. The codec has been implemented in real time on two Tiger C40 boards. Informal listening tests using this real-time hardware seem to indicate that this candidate is in close proximity to the full rate standard. >
4 citations
19 Apr 1994
TL;DR: A low bit-rate speech codec has been developed and the speech coder employs a PCELP (CELP with pulse codebook) algorithm, which uses a pulse-train excitation codebook to enhance the quality of voiced speech.
Abstract: A low bit-rate speech codec has been developed. The speech coder employs a PCELP (CELP with pulse codebook) algorithm, which uses a pulse-train excitation codebook to enhance the quality of voiced speech. The computational complexity for the pulse codebook search is greatly reduced by using truncated impulse responses of a weighted synthesis filter. An improved Viterbi decoding method using error detecting code information instead of tail bits has also been developed to increase the effective error-correction coding rate. A 5.6 kb/s speech codec has been implemented by incorporating the PCELP speech coding algorithm with the improved Viterbi decoding. Formal listening tests show that the synthetic speech quality of this codec is equivalent to that of 5.4-bit /spl mu/-law PCM in the absence of channel errors. >
2 citations
03 Aug 1994
TL;DR: The requirements and selection criteria of software C ODEC algorithms in DVC are presented, and an overview of CODEC algorithms for DVC, such as Differential Pulse Code Modulation (DPCM), Discrete Cosine Transform (DCT), Wavelet, Vector Quantization, Fractal, Block Truncation Coding and Hybrid coding are presented.
Abstract: The computational power of desktop computers (e.g. Intel 486 PC) has reached the stage where software solutions for desktop videoconferencing (DVC) are worth considering. Here, coder/decoder (CODEC) refers to video compression/decompression. In this paper, the requirements and selection criteria of software CODEC algorithms in DVC are presented. An overview of CODEC algorithms for DVC, such as Differential Pulse Code Modulation (DPCM), Discrete Cosine Transform (DCT), Wavelet, Vector Quantization, Fractal, Block Truncation Coding and Hybrid coding will give us insight into their feasibility in DVC, particularly when implemented in software. Some of the driving factors in the proliferation of DVC are: cost, platform, interoperability, communications bandwidth, availability and integrity, performance and quality, and last but not the least, collaborative applications. Most of these factors, especially interoperability, are important in choosing a software CODEC algorithm and so the standard H.261 is worth paying attention to. A quantitative analysis of some CODEC techniques is presented in a matrix.
1 citations