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.

Waveform interpolation for 4 kbit/s wideband speech coding

Abstract: An implementation of waveform interpolation (WI) applied to wideband speech coding at 4 kbit/s is presented. Listening tests comparing narrowband speech coded using standard coders are presented, with wideband speech coded using WI. Results show a clear preference for wideband speech over narrowband speech coded at an equivalent bit rate.

Three-Stage Iterative Detection of a MIMO-Aided Precoded AMR-WB Speech Transceiver

Abstract: Ajointlyoptimisediterativesource and chan- nel decoding (ISCD) scheme invoking the Adaptive Multi RateWideband (AMR-WB)speech codec isproposed. More explicitly, the transceiver investigated consists of serially concatenated Recursive Convolutional (RSC)codes, aUnity Rate Code (URC) referred to as a precoder and the AMR- WB speech codec, where the resultant bitstream is trans- mittedusingDifferential Space-Time Spreading (DSTS) and Sphere Packing (SP) modulation over narrowband tempo- rallycorrelated Rayleigh fading channels. Theconvergence behaviour of the advocated scheme is investigated with the aid of Extrinsic Information Transfer (EXIT) charts, which suggests that potential performance improvements can be achieved by combining a serially concatenated precoder with the AMR-WB speech codec. The proposed system exhibits an Eb/N0 gain of about 1.5 dB in comparison to the benchmark scheme carrying out iterative source- and channel-decoding as well as DSTS aided SP-demodulation, but dispensing with the precoder, when using Isystem =4 system iterations.

Stability Analysis of Speech Synthesis Filter of CELP-Based AMR-WB Codec

Abstract: The speech codec analyzes the speech using A(z) (analysis filter) and synthesizes back at decoder side using linear prediction coefficients (LPC). These LP coefficients are sensitive and cannot be sent directly in a transmission channel. A small corruption in LPC values during transmission destroys the synthesized speech at the decoder side. We have presented new results on the stability and sensitivity of LPC based on changes in speech input pitch length, sign bit, and LPC values during transmission (or for any other reason) consecutively and simultaneously. Present analysis will help to add varying dynamic range to LSF coding. For this each individual LPC need to be related to each LSF. All the speech inputs considered in this study are voiced speech, which has been separated manually. For a specific order, we analyzed the numbers of LPC which are more responsible for increase in prediction error at decoder side when they are corrupted by noise. Present analysis provides the reference for number of bits required for quantization of LPC or line spectral pairs (LSF).

Performance comparison of audio codecs for high-quality color ring-back-tone services over CDMA

Abstract: In this paper, we investigate the use of existing audio codecs for the purpose of a high quality color ring-back- tone service. First of all, we exploit the limitations of the enhanced variable rate codec (EVRC) in a view of music quality because EVRC is a standard speech coder employed in a code division multiple access (CDMA) system. In order to figure it out which current existing audio codec is suitable to deliver music over CDMA or wideband CDMA (W-CDMA), several audio codecs such as two different versions of MPEG AAC and the Enhanced AAC+ codec are reviewed. Next, the music quality of the audio codecs is compared with that of EVRC, where the bit-rates of the audio codecs are set to be around 10 kbit/s because the color ring-back-tone service using one of the audio codecs should be realized by replacing EVRC with it. The quality comparison is performed by an informal listening test as well as an objective quality test. It is shown from the experiments that the audio codecs provide better music quality than EVRC and among them, the Enhance AAC+ codec operated at a bit-rate of 10 kbit/s with a sampling rate of 32 kHz can be considered as a new candidate for the high quality color ring-back-tone service.

Rate Quality Factor Optimization for VP8 Video Codec for Mobile Communication Applications

Abstract: In this paper, mainly, we are proposing a better rate control algorithm optimization technique; it is applying for VP8 video coding for improving the performance and also quality of video codec data for mainly mobile communication applications. Actually, rate control plays an better role in video coding and transmission to provide the best video quality at the receiver end, and our proposed algorithm technique mainly exploits the existing constant quality control, which is governed by a parameter called quality factor (QF) to give a constant bit rates. For this purpose, a new mathematical model called the rate–quality factor(R–Q′) is derived to generate optimum QF for the current coding frame using the bitrates resulting from the encoding of the previous frame in order to meet the target bitrates. The process of calculating the QF is simple, and further calculation is not required for each coded frame. It also provides the rate control solution for both intra-frame-only and inter-frame coding modes. Our all experimental results show that the proposed scheme generates coding bits very close to target bits and provides improved coding efficiency at low bit rates also introducing simple complexity to measure for the video content. So that, in this proposed method considering previous results and comparing with our results then confirm by comprehensive experiments results.. The quality control parameter can be derived from Lagrangian multiplier and hence can be used in any type of encoder that uses RDO. Experimental results have shown that the proposed algorithm can control the bitrates within 1 % of the target bitrates on average, and it has better bitrates regulation over each GOP than the rate control algorithm of H.264. This is an advantage that is crucial in real-time multimedia data streaming in preventing buffer overflow or underflow. Performance of this video data is analyzed using PSNR and bitrates which needs to calculate once this new technique is implemented on specified plat form.