Audio Coding Algorithms Research Articles

The Influence of the Bitrate Level on the Subjective Sound Quality Perception of the Concatenated Non-Entropic Audio Coding Algorithms in the Digital Broadcasting Chain

The Influence of the Bitrate Level on the Subjective Sound Quality Perception of the Concatenated Non-Entropic Audio Coding Algorithms in the Digital Broadcasting Chain

高能率符号励振線形予測符号化と帯域選択型変形離散余弦変換符号化を用いた32kbit/sスケーラブル広帯域音声・オーディオ符号化方式の開発

高能率符号励振線形予測符号化と帯域選択型変形離散余弦変換符号化を用いた32kbit/sスケーラブル広帯域音声・オーディオ符号化方式の開発

English

In this paper we present a new lossless audio coding algorithm using Burrows-Wheeler Transform (BWT) and Run Length Encoding (RLE).Audio signals used are assumed to be of floating point values. The BWT is applied to the audio signals to get the transformed coefficients and then these resulting coefficients are better compressed using Run Length Encoding. Two entropy coding are used which are Run Length Encoding and Huffman coding. Proposed compression algorithm is experimented and analyzed for two different stereo type audio signals. Compression ratio and Bit rate for audio coding has been used as a comparison parameter for proposed audio coding algorithm. Experimental result shows that the lossless audio coding algorithm outperforms other lossless audio coding methods; using combined Burrows Wheeler Transform & Move to front coding method ,using combined Burrows Wheeler Transform and Huffman coding method, and using Burrows Wheeler Transform ,Move to front coding method & Run Length Encoding method.

Bandwidth Extension for Hierarchical Speech and Audio Coding in ITU-T Rec. G.729.1

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

Huffman code based error screening and channel code optimization for error concealment in perceptual audio coding (PAC) algorithms

The class of perceptual audio coding (PAC) algorithms yields efficient and high-quality stereo digital audio bitstreams at bit rates from 16 kb/sec to 128 kb/sec (and higher). To avoid "pops and clicks" in the decoded audio signals, channel error detection combined with source error concealment, or source error mitigation, techniques are preferred to pure channel error correction. One method of channel error detection is to use a high-rate block code, for example, a cyclic redundancy check (CRC) code. Several joint source-channel coding issues arise in this framework because PAC contains a fixed-to-variable source coding component in the form of Huffman codes, so that the output audio packets are of varying length. We explore two such issues. First, we develop methods for screening for undetected channel errors in the audio decoder by looking for inconsistencies between the number of bits decoded by the Huffman decoder and the number of bits in the packet as specified by control information in the bitstream. We evaluate this scheme by means of simulations of Bernoulli sources and real audio data encoded by PAC. Considerable reduction in undetected errors is obtained. Second, we consider several configurations for the channel error detection codes, in particular CRC codes. The preferred set of formats employs variable-block length, variable-rate outer codes matched to the individual audio packets, with one or more codewords used per audio packet. To maintain a constant bit rate into the channel, PAC and CRC encoding must be performed jointly, e.g., by incorporating the CRC into the bit allocation loop in the audio coder.

A combined wideband speech and audio coder using human articulatory and auditory models

The modern high-quality speech and audio coding algorithms are mostly based on human articulatory models. On the other hand, high-quality audio coders are mostly based on models of human auditory system. However, neither of these two methods can provide acceptable performance for both music and speech. In this paper a coder is proposed which operates for both wideband speech and audio signal. The structure of the coder is based on both human articulatory and auditory models. In the first part the LPC parameters are drawn using the conventional autocorrelation method in order to simulate the human articulatory system. With LPC parameters the redundancy of input signal in time domain is removed. In the second part of the coder, the LPC residual signal is further transformed into frequency domain through MLT (modulated lapped transform). The MLT coefficients are then quantized in frequency domain using the human auditory properties such as masking property. The informal listening test has shown that the proposed coder provides very good quality for wideband speech and for most music signals.

Cycle saving hardware for real-time audio processing

The authors present a design for cycle saving hardware components that enable real-time operation of audio coder-decoders. To determine the key components of audio coding algorithms, the MPEG2 audio algorithm is analysed, which is an international standard and uses a mixture of several coding strategies. Through the analysis, some components are selected that significantly contribute to the total number of cycles, and corresponding hardware accelerators that complete assigned tasks in a single cycle are designed. By incorporating these accelerators, a simple processor can reduce the number of cycles for assigned tasks by up to 72.2%.

Auditory analysis of spectro-temporal information in acoustic signals

Acoustic signals are often characterized by pitch, timbre, loudness, forms of modulation, and onset/offset instants. These descriptors of sound quality have a close relationship to the instantaneous spectral properties of the sound waves. The auditory system has developed elegant mechanisms to extract and represent this spectro-temporal information through noise-robust perceptual features. After decades of research, the anatomical organization and functional significance of many of the underlying mechanisms are now becoming understood. Consequently, mathematical descriptions of the various early and central auditory stages have become feasible and functionally relevant, and can be fruitfully used to derive principles underlying time-frequency analysis in the auditory system. Here, the authors review examples of such approaches from their recent efforts to model the early and central stages of the mammalian auditory systems. It is within the hope of this study to better understand the nature of hearing, and to eventually apply these functional principles to designing engineering applications, such as audio coding algorithms that are closely optimized under perceptual criteria, and automatic speech recognition systems that have performance comparable to humans.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Visual telephony as an ISDN application

The key factors deterring the use of visual telephony are identified, and an overview of a typical system architecture is given. The video signal formats and video and audio coding algorithms used are described. Video codec implementation is considered, and an implementation based on application-specific integrated circuits is presented. In particular, three key signal processing modules in the video codec are examined: a discrete cosine transform chip, a motion estimation chip, and a variable-length codec chip. Standardization activities in the video coding area are discussed. >