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.

Adaptive audio codec system, method and article

Abstract: An encoder includes a low-pass filter to filter input audio signals. The low-pass filter has fixed filter coefficients. The encoder generates quantized signals based on a difference signal. The encoder includes an adaptive quantizer and a decoder to generate feedback signals. The decoder has an inverse quantizer and a predictor. The predictor has fixed control parameters which are based on a frequency response of the low-pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low-pass filter and the encoder. The adaptive noise shaping filter flattens signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.

A Study on ACFBD-MPC in 8kbps

Abstract: Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

On the applicability of the SBC codec to support super-wideband speech in Bluetooth handsfree communications

Abstract: With the recent standardization of the Enhanced Voice Services (EVS) codec in 3GPP, mobile operators can upgrade their voice services to offer super-wideband (SWB) audio quality (with 32 kHz sampling rate). There is however one important use case which is currently limited by existing standards: hands free communication with wireless headsets, car kits, or connected audio devices often rely on Bluetooth, and the hands free-profile (HFP) in Bluetooth is currently limited to narrowband and wideband speech. Following the approach used to extend HFP to support wideband, we study in this paper the applicability of the SBC codec to further extend HFP to SWB. An evaluation of performance is provided taking into account Bluetooth system constraints.

An audio coding system using subband backward linear prediction

Abstract: In this paper, block backward adaptive linear predictors are used for improving the performance of perceptual audio codecs. Based on the investigation on different linear prediction algorithms, a signal dependent adaptive-switched predictor is developed. This predictor not only delivers significant coding gain for stationary signals but also recovers quickly from transient signals. At a bitrate of 64 kbit/s, the performance of the new codec for most critical test sequences is significantly better than MPEG-1 Layer II.