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.

Joint Enhancement and Coding of Speech by Incorporating Wiener Filtering in a CELP Codec.

Abstract: The performance of speech communication applications in the field of mobile devices is often hampered by background noises and distortions. Therefore, noise attenuation methods are commonly used as a pre-processing method, cascaded with the speech-codec. We demonstrate that the performance of such combinations of speech enhancement and coding methods can be improved by joining the two methods into a single block. The proposed method is based on incorporating Wiener filtering into the objective function used for optimization of the quantization in code excited linear prediction (CELP)-based codecs. The benefits are that 1) the non-linear components of CELP codecs, including quantization and error feedback, are taken into account in the joint minimization function thereby improving quality and 2) by merging blocks both delay and computational complexity can be minimized. Our experiments demonstrate that the proposed joint enhancement and coding approach consistently improves subjective and objective quality. The proposed method is compatible with any CELP-based codecs without changing the bit-stream, whereby it can be readily applied in mobile phones or speech communication devices applying the concepts of CELP codecs for improving perceptual quality in adverse conditions.

MPEG-2 real-time software CODEC for full-duplex transmission application over IP networks

Abstract: This paper proposes a real-time MPEG-2 software CODEC for full-duplex transmission applications, and evaluates its performance and usefulness. The CODEC consists of a high-speed encodersdecoder, an IP sendersreceiver, and an error recovery controller. Each encodersdecoder is accelerated and optimized by exploiting fast algorithms and instruction-level parallelism. The IP sendersreceiver combination achieves low delay owing to the direct translating of each elementary stream of video and audio into UDPsIP packets. The error recovery controller carries out simple but powerful error tolerance against packet loss over IP networks. This CODEC attains low delay of 99 ms (M = 1, N = 1) to 165 ms (M = 3, N = 3) including input, encoding, transmitting, decoding, and output delays, and maintains a normal frame rate of 30 fps (frames per second) and more than 20 fps even under a fairly heavy network load. It provides sufficiently good performance for use as a real-time full NTSC-size CODEC on a PC of at least 1.2-GHz CPU. © 2005 Wiley Periodicals, Inc. Syst Comp Jpn, 36(2): 33–41, 2005; Published online in Wiley InterScience (). DOI 10.1002sscj.20151

Classification of audio codecs based on coding characteristics

Abstract: We propose a fast, simple and new method for identification of audio codecs that does not require decoding of coded audio data. The method uses chaotic and randomness features of coded audio to build models associated with different codecs. The most important features of the method are operating on just a few kilobytes of data sampled randomly from audio file and assuming no knowledge on encoding structure of a codec. These two features make it fast and simple. Experiments are performed on both singly coded and transcoded audio samples to measure accuracy of the technique.

Super-wideband fine spectrum quantization for low-rate high-quality MDCT coding mode of the 3GPP EVS codec

Abstract: This article presents a low bit-rate super wideband MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance at 13.2 kbps, existing algorithms are reviewed and several new tools are introduced into the low bit-rate MDCT coder to improve the performance of the coder while coding music and mixed content. A subjective listening test demonstrates the advantage of the proposed system for 13.2 kbps when compared to AMR-WB+.

Research on audio bandwidth extension codec

Abstract: With stereo audio stepping into people's daily lives, spatial parametric technology comes to its broad prospects for development. This paper provides the audio bandwidth extension encoding and decoding devices, the bitrate of less high-frequency signal reconstruction, in order to improve the quality of the output audio signal. This research makes a contribution to the lack of existing technology, audio bandwidth extension encoding and decoding devices, digital audio decoding to take advantage of a lower bit rate to restore the original audio signal in the digital encoding process the loss of high-frequency components, thereby enhancing the sound playback quality.