Kazunori Ozawa

Speech coding/decoding method having an excitation signal

A speech coding method in which spectrum parameter representing a spectrum envelope and a pitch parameter representing a pitch are obtained from an input discrete speech signal. A frame interval is divided into subintervals in accordance with the pitch parameter. A sound source signal in one of the subintervals is obtained by obtaining a multipulse with respect to a difference signal obtained by performing prediction on the basis of a past sound source signal. Correction information for correcting at least one of the amplitude and the phase of the sound source signal are obtained and output in other pitch intervals in the frame.

A bitrate and bandwidth scalable CELP coder

This paper proposes a flexible CELP speech coder with bitrate and bandwidth scalabilities for multimedia applications. The coder is based on multi-pulse-based CELP coding and consists of a bitrate scalable base-band coder and a bandwidth extension tool. The bitrate scalable base-band CELP coder employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. The bandwidth scalability is realized by bandwidth conversion from base-band CELP parameters to those for wideband without a widely used subband structure. The bandwidth conversion improves base-band coding quality and expands bandwidth, simultaneously. The comparison test results show that the bitrate scalable coder is equivalent in speech quality to the fixed-bitrate CELP coder at the same bitrate for the narrowband speech. In the mean opinion score (MOS) tests, the proposed 16 kbit/s coder with the bandwidth scalability achieves equivalent coding quality to ITU-T G.722 at 56 kbit/s. The proposed coder is currently evaluated as the MPEG-4 CELP speech standard.

Speech analysis and synthesis system

A speech analysis and synthesis system operates to determine a sound source signal for the entire interval of each speech unit which is to be used for speech synthesis, according to a spectrum parameter obtained from each speech unit based on cepstrum. The sound source signal and the spectrum parameter are stored for each speech unit. Speech is synthesized according to the spectrum parameter while controlling prosody of the sound source signal. The spectrum of the synthesized speech is compensated through filtering based on cepstrum.

Design principles of an operator-owned highly distributed content delivery network

Mobile network operators are experiencing a tremendous increase in data traffic due to the growing popularity of bandwidth-intensive video services. This challenge can be faced either by boosting the capacity of the network infrastructure, or by means of offloading traffic from the backhaul and core network and serving contents from distributed cache servers close to the users. Network operators can extend the coverage of traditional CDNs by making usage of caching locations much closer to the users than traditional CDNs. Additionally, network operators can optimize the caching and delivery of contents by exploiting the complete knowledge of their network for designing a cost-effective infrastructure able to achieve both improved user satisfaction and cost savings. This article provides thoroughly justified design principles for a highly distributed operator-owned CDN while focusing on four key aspects: the optimal location of cache servers, mechanisms for request routing, content replica placement, and content outsourcing and retrieval.

Speech decoder for high quality reproduced speech through interpolation

A speech decoder includes a separating circuit, an error correction decoding circuit, an interpolating circuit, and a speech reproducing circuit. The separating circuit separates a code string of a filter parameter, a code string of a parameter associated with a pitch, and a code string of a parameter associated with an index and a gain of a codebook representing an excitation signal of speech from a received code string. The error correction decoding circuit detects a transmission error, which cannot be corrected, in the received code string. When a transmission error which cannot be corrected is detected, the interpolating circuit interpolates between parameters of past and future proper frames, thereby recovering parameters of a current frame. The speech reproducing circuit reproduces a speech signal on the basis of the interpolated parameters and other received codes.

M-LCELP speech coding at 4 kbps

This paper presents the M-LCELP (multi-mode learned code excited LPC) speech coder, which has been developed for the North American half-rate digital cellular systems. M-LCELP develops the following techniques to achieve high-quality synthetic speech at 4 kbps: (1) Multimode and multi-codebook coding, (2) Pitch lag differential coding with pitch tracking, (3) A two-stage joint design regular-pulse codebook with common phase structure in voiced frames, (4) An efficient vector quantization for LSP parameters, (5) An adaptive MA type comb filter to suppress excitation signal inter-harmonic noise. The MOS subjective test shows that 4.075 kbps M-LCELP synthetic speech quality is high, and that its quality is mostly equivalent to that for an 8 kbps North American full-rate VSELP coder.<<ETX>>

A 32 kb/s toll quality ADPCM codec using a single chip signal processor

An ADPCM codec, that can provide toll quality speech at a 32 kb/s transmission rate, has been implemented on a single chip signal processor. Maximum effort has been paid to design a robust adaptation scheme for a quantizer and a predictor to withstand transmission bit errors. The codec employs a simplified robust quantizer and also employs a new backward adaptive predictor. The decoder, including the new adaptive predictor, has a structure having fixed poles and adaptive zeros, attaining both high prediction capability and robustness. The performance of a developed codec, which has analog interface capability through a PCM codec chip, satisfies the standard 64 kb/s PCM performance specification in CCITT recommendation G.712.

A Study on Pulse Search Algorithms for Multipulse Excited Speech Coder Realization

This paper describes and compares several kinds of pulse search methods for multipulse excited speech coder realization. These pulse search methods are derived from minimization criterion for errorpower between original speech and synthetic speech, but their performances and required computation amounts are different. Objective and subjective evaluations are carried out to compare the performances. Further, the relation between speech quality and pulse search method complexity is described. Based on these results, pulse search methods suitable for realizing a high-quality multipulse coder using current VLSI technology are discussed.

An adaptive multi-rate speech codec based on MP-CELP coding algorithm for ETSI AMR standard

This paper proposes a speech codec based on the multi-pulse based CELP (MP-CELP) coding and convolutional coding algorithms for the ETSI adaptive multi-rate (AMR) standard. The codec operates at several speech coding rates, maintaining a fixed gross rate including speech and channel coding for the full-rate (FR) and half-rate (HR) channel modes. MP-CELP has great features of easily changing the speech coding rate by controlling the parameters such as the number of pulses and other parameters. Subjective tests show that the proposed AMR codec in the FR channel mode achieves higher performance than that of the enhanced FR codec, and the proposed codec in the HR channel mode gives a comparable coding quality to that by the full-rate codec, by selecting an optimal coding rate for each channel condition. T-tests based on the test results also show that the proposed speech codec meets about 80% of the seventeen requirements, which are selected from the AMR standard study report. Therefore, the proposed codec is promising for the AMR standard.

A post-processing technique to improve coding quality of CELP under background noise

This paper proposes a novel post-processing technique to improve the coding quality of CELP under background noise. It adaptively smoothes both the spectral envelope and the energy of the estimated excitation signal to reduce their temporal fluctuations, which cause the perceptual degradation. The excitation signal is calculated using the synthesized signal and the spectral parameters given from the decoder. Thus, the proposed post-processing is performed separately from the decoder. The smoothing is applied only in non-speech periods and the smoothing strength is controlled depending on the characteristics of the synthesized signal to avoid the degradation in speech and non-stationary noise periods. Subjective test results show that the proposed post-processing improves degradation mean opinion score (DMOS) by 0.2 to 0.4 for noisy speech signals, which are coded by the GSM adaptive multi-rate (AMR) codec.