Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

A low-bit-rate video codec using block-list-transform

Abstract: A simple but efficient video codec is built for low-bit-rate videophone applications. The goal of this project is to construct a video codec at a low cost and good performance. In order to reduce hardware complexity, a simple interframe predictive coding structure is selected. The input pictures are partitioned into 2-D blocks, and only the blocks with significant frame differences are coded and transmitted to the decoder. The Block-list-transform (BLT) coding algorithm, a generalized DPCM, is used to encode the frame differences between the previously reconstructed picture and the current picture. Preliminary results using this coding system at 64 kbits/sec show reasonable coding performance on typical videophone sequences.

Speech Coding Using Least-Squares Estimation

Abstract: An important goal in current speech coding research is providing high-quality speech at low bit rates (4.8–16 Kbps). Several methods [1]–[3] have been proposed recently to achieve this end. Compared to the conventional linear predictive (LP) vocoder [4], these methods employ an enhanced speech production model to synthesize speech. For example, instead of a single stage, the modulation filter now typically consists of two stages: i) a short-delay filter modeling the spectral envelope of speech, and ii) a long-delay filter modeling the spectral fine structure. Both are time-varying, all-pole filters and are derived from the original speech through LP analysis. Also, some information is provided about the excitation signal, which is selected by means of an analysis-by-synthesis procedure whereby a perceptually weighted error criterion is minimized In the multi-pulse linear predictive coder (MPLPC) [1], the excitation signal is a sequence of appropriately located and scaled impulses. In the code excited linear predictive coder (CELPC) [2], it is an entry from a codebook of white, gaussian noise sequences. In the self excited vocoder (SEV) [3], it is selected from the past history of the source excitation. As a result of these improvements, the above coders are able to synthesize high-quality speech at low bit rates.

Performance enhanced scalable wideband speech coding for IP networks

Abstract: The scalable wideband speech coding scheme based on the internet low bitrate codec (iLBC) was previously presented and achieved speech quality equivalent to ITU-T G.729.1 at high bit rates. However, the performance was limited at low bit rates. In this paper, we present various approaches to improve performance especially at low bit rates. In particular, the time-domain bandwidth extension (TDBWE) is employed for higher-band coding, and the efficient coding structure is employed in enhancement layers. The performance evaluation results show that significant improvement is achieved at low bit rates and the proposed codec outperforms G.729.1 at most bit rates.

Encoding and error correction system for enhanced performance of legacy communications networks, and method thereof

Abstract: An encoding and error correction system and method employs an AMR codec (18) by stripping header data from a plurality of legacy system frames (10) having header and traffic channel (TCH) data blocks.Speech data is then encoded using the AMR to create bits for a data block substantially the same as contained in the plurality of frames. The stripped header data is encoded as a long frame header using a fixed convolution coder(24). The speech data is then convolutionally encoded and the long frame header and encoded speech data are combined as a long frame (32). The long frame is then deconstructed into a plurality of equal segments (106, 110) and the segments are transmitted as TCH data in the legacy system frame format.

Method, server device and a converting device for setting up a user data connection

Abstract: According to the invention, in order to transmit user data from a source communications device (KE1) provided with a first encoder (CODEC_A, CODEC_B1, CODEC_C) for encoding users data to a target communications device (KE2) provided with a first decoder (CODEC_J2, CODEC_K) for decoding said user data via a communication network which is provided with several converting devices (GW1, GW2, GW3) comprising additional encoders (CODEC_B2, CODEC_D, CODEC_H2,...) and additional decoders (CODEC_B2, CODEC_D, CODEC_H2,...) for carrying out the verification of the converting devices (GW1, GW2, GW3). Said verification consists in determining, whether the first encoder (CODEC_A, CODEC_B1, CODEC_C) is compatible with the decoder (CODEC_B2, CODEC_D, CODEC_H2,...) of a given converting device (GW1, GW2, GW3) and, whether the first decoder (CODEC_J2, CODEC_K) is compatible with the encoder (CODEC_B2, CODEC_D, CODEC_H2,...) of said converting device (GW1, GW2, GW3). One of the converting devices (GW1, GW2, GW3) for which the compatibility is ascertained by the verification is selected for transmitting user data. During transmission of the user data, said user data encoded with the aid of the first encoder (CODEC_A, CODEC_B1, CODEC_C) is decoded with the aid of the compatible decoder (CODEC_B2, CODEC_D, CODEC_H2,...) of the converting device (GW1, GW2, GW3) and the user data decodable with the aid of the first decoder (CODEC_J2, CODEC_K) is encoded with the aid of the compatible encoder (CODEC_B2, CODEC_D, CODEC_H2,...) of the selected converting device (GW1, GW2, GW3).