Showing papers on "Enhanced Variable Rate Codec published in 1999"

The Enhanced Variable Rate Coder: Toll quality speech for CDMA

Abstract: The Enhanced Variable Rate Coder (EVRC), standardized by the Telecommunications Industry Association (TIA) as IS-127, is intended for use with the IS-95x Rate Set 1 air interface (CDMA) This coder operates at a maximum rate of 85 kb/s and an average rate of about 41 kb/s on conversational speech The EVRC consists of three coding modes that are all based on the Code Excited Linear Prediction (CELP) model Selection among the three modes is based on an estimate of the input signal state, with active speech encoded primarily at 170 bits/20 msec frame (Rate 1), background noise and silence encoded at 16 bits/frame (Rate 1/8), and some active speech and essentially all transitions between speech and silence encoded at 80 bits/frame (Rate 1/2) In order to improve performance in the presence of background noise, the EVRC employs an adaptive noise-suppression filter at the input Subjective test results are presented demonstrating that the EVRC delivers excellent quality voice in clean speech/clear channel conditions, and that its performance exceeds that of most currently standardized speech coders for wireless applications in background noise and/or impaired channel conditions

Method for eliminating annoying noises of enhanced variable rate codec (EVRC) during error packet processing

Abstract: A decoding method of an Enhanced Variable Rate Codec (EVRC) is provided for reducing EVRC noises during error packet processing in a CDMA system. The method induces setting a seed value for generating background noise when the input error packet is a first error packet, and the seed value is maintained when the error packet is inputted continually. In addition, an average fixed codebook gain and an average eighth gain are decayed, so that the decoder of the EVRC can generate more comfort background noises without loud annoying noises. Further, an excitation signal is post filtered selectively, so that granular noise can be prevented and speech quality of the decoder output is improved.

A method and apparatus for efficient bandwith usage in a packet switching network

Abstract: A universal CODEC, in one implementation, comprised of multiple CODECs in a bank, each CODEC capable of encoding/decoding speech transmissions in a different format, replaces a customary installation of two CODECs in a BSC/MSC. Within the universal CODEC, a bypass feature is incorporated that is capable of automatically selecting the individual CODEC having a format that is appropriate to a received signal. A BSC/MSC with a universal CODEC may receive any encoded voice signal and convert to the format appropriate to the receiving BSC/MSC. A second implementation of the universal CODEC utilizes multiple software applications and shares a common digital signal processor (DSP) core. Utilizing an automatic bypass feature and low frequency, in-band signals, the universal CODEC provides means to consume less bandwidth while transmitting speech signals only in compressed format through packet switching networks.

Multiple technology vocoder and an associated telecommunications network

Abstract: A wireless telecommunications network includes a base station controller which, in turn, is comprised of a call control resource manager and a multiple technology vocoder having first and second interfaces, first and second selection managers, a vocoder controller and first, second, third, fourth and fifth resources, each embodied as a software module, for performing D/A and A/D conversions on messages configured in accordance with EVRC, EFRC, RLP, VSELP and QCELP protocols, respectively. When transferring messages between a mobile terminal and an MSC via the base station controller, the mobile terminal first informs the call control resource manager of the protocol to which the message conforms. In turn, the call control resource manager advises of the vocoder controller of the protocol type for the incoming message. The vocoder controller configures the interfaces to accept messages formatted in the protocol and instructs the selection managers to provide a path, for the received message, to the appropriate resource for conversion. The converted message is then returned to the call control resource manager and transmitted on to its final destination.

The MSM3000/sup TM/ dual-mode CDMA+AMPS ASIC: for increased functionality and standby time of handsets

Abstract: The MSM3000 dual mode ASIC for digital baseband processing within CDMA cellular handsets is a significant leap forward in functionality. The improvements include an upgraded microprocessor, several standby time improvements, an upgraded vocoder DSP to allow for cost effective EVRC, and support for multiple code channel reception for packet data service speeds up to 86.4 kbps in the forward direction per new TIA standards.

A low bit rate codec for AMR standard

Abstract: We describe a low bit rate speech codec based on the RCELP paradigm and designed as a candidate for GSM-AMR. The relaxation of the waveform-matching constraint in the RCELP model allows for reducing the bit rate without affecting the speech quality. New efficient quantization methods for the LSF and gain parameters coupled with some algorithmic improvements result in a high quality speech codec at bit rates as low as 4.55 kbit/s. Subjective tests show encouraging results in terms of quality and robustness under various operating conditions.

An improved background noise coding mode for variable rate speech coders

Abstract: In this paper, we present a novel background noise coding scheme for variable rate speech coders. Existing approaches to noise coding at very low bit rates (i.e. below 1 kbps) fail to faithfully reproduce background noise resulting in a degradation of the overall perceptual quality. In our approach, classification of the noise type is used to select the type of excitation to be used at the receiver. To illustrate the benefits of our scheme, we have modified the noise coding mode of the CDMA enhanced variable rate codec (EVRC) to include the proposed class-dependent noise excitation model. Evaluation tests have shown that we have improved the overall quality with the proposed noise coding scheme without an increase in bit rate.

Rate control of a region-based H.263 video codec under time-varying channels

Abstract: A region-based video codec based on the H.263+ standard is examined and its associated novel rate control schemes are proposed for visual communication through time-varying low-bit-rate channels. The region-based coding scheme is a hybrid method that consists of the traditional block DCT coding and the object-based coding. Basically, we adopt H.263+ as the platform, and develop a fast macroblock-based segmentation method to implement the region-based video codec. The proposed rate control solution includes rate control in three levels: encoding frame selection, frame-layer rate control and macroblock-layer rate control. The goal is to enhance the visual quality of decoded frames without obvious motion unsmoothness under time-varying channels. The efficiency of the proposed rate control schemes applied to the region-based video codec is demonstrated via several typical test sequences.

Improving EVRC half rate by the algebraic VQ-CELP

Abstract: This paper presents an algebraic vector quantized codebook excited linear prediction (AVQ-CELP) speech codec. The objective is to enhance the half rate mode of IS-127, the enhanced variable rate codec (EVRC). In AVQ-CELP scheme, only the perceptually important components are encoded, and the selection of the components is done in a way similar to the ACELP. An open-loop procedure is used to select the subvectors. The selected sub-vectors are concatenated and vector quantized. An analysis-by-synthesis strategy is used to determine the optimal excitation. The generalized Lloyd algorithm (GLA) is used to optimize the AVQ codebook. In order to improve the synthesis quality of voiced frames, a two-pulse version of ACELP is used in the strong voiced frames. The proposed algorithm was incorporated in the Nokia CDMA handset prototype. Under a joint collaboration effort with SK Telecom, a field-testing was performed in Korea to evaluate the performance of the proposed AVQ algorithm. The results indicate a considerable improvement relative to the standard EVRC operating at the maximum half-rate.

Enhancing the EVRC half rate by the algebraic VQ-CELP

Abstract: This paper presents some update improvement in an algebraic vector quantized codebook excited linear prediction (AVQ-CELP) speech codec. The objective is to enhance the half rate mode of the enhanced variable rate codec (EVRC). In the AVQ-CELP scheme, only the perceptually important components are encoded, and the selection of the components is done in a way similar to the ACELP. A closed-loop procedure is used to select the sub-vectors. The overlapping between the selected vectors are allowed to prevent the pitch peak splitting. The selected sub-vectors are concatenated and vector quantized. An analysis-by-synthesis strategy is used to determine the optimal excitation. The generalized Lloyd algorithm (GLA) is used to optimize the AVQ codebook. In order to improve the synthesis quality of voiced frames, ACELP is used in the strong voiced frames. The proposed algorithm was incorporated in the Nokia CDMA handset prototype. The field testing results indicate a considerable improvement relative to the standard EVRC operating at the maximum half-rate.

A multi-rate speech and channel codec: a GSM AMR half-rate candidate.

Abstract: The current GSM standard is a fixed rate system which has been optimised to give good performance in all channel conditions. However since the channel conditions in a terrestrial mobile communication network such as GSM vary significantly between the best and the worst case, the existing GSM speech and channel coding performance can be improved by incorporating the dynamic channel conditions into the codec design. Under the initiative of the European Telecommunications Standards Institute (ETSI) such a system has been launched. This new standard is called AMR for Adaptive Multi-Rate: the source and channel coding rates can be adapted depending on the state of the channel, thus providing optimal balance between them at any time. The University of Surrey has submitted a candidate for this competition through the Mobile VCE. This candidate was the only one amongst eleven to use a vocoder in the half-rate GSM channel instead of a CELP based coder and tests ranked it among the best. This paper presents the system submitted for the half-rate channel as well as the results of the testing.