Performance Evaluation of Multichannel Audio Compression

Instrumentation and Measurement Technology Conference, 1996. IMTC-96. Conference Proceedings. Quality Measurements: The Indispensable Bridge between Theory and Reality., IEEE, 1996

One of the most difficult issues about audio-codec design and evaluation is the analysis of the reproduced sound quality. Standard measurements (such as the signal-to-noise ratio) aren't adequate to make an accurate comparison between different codecs due to the masking phenomena of the human ear. The aim of this paper is to propose a simple objective test for the measurement of the coding distortion in analysis-by-synthesis codecs

IEEE Transactions on Audio, Speech, and Language Processing, 2013

International Journal of Computer Applications, 2014

The paper presents a comparative study of audio compression using multiple transformation techniques. Audio compression with different transform techniques like Discrete Cosine Transform, Wavelet Transform, Wavelet Packet Transform (W.P.T) & Cosine Packet Transform is analyzed and compression ratio for each of the transformation techniques is obtained. Mean Compression ratio is calculated for all of the techniques and compared. Performance measures like signal to noise ratio (SNR), normalized root mean square error (NRMSE), retained signal energy (RSE) are also calculated and compared for each transform technique. Transform based compressed signals are encoded with encoding techniques like Run-length Encoding (R.L.E) and Mu-Law Encoding to reduce the redundancies. From the comparison it is clear that Discrete wavelet transform gives better compression ratio of about 27.8593 compared with the other three transforms. Mean SNR value is minimum for DCT 29.2830, and comparatively higher mean SNR value 43.4037 for CPT.

The Journal of the Acoustical Society of America, 2000

This paper provides a subjective quality analysis of transforms used in audio compression algorithms for a class of music signals. A 34-subject listener test compares three transforms in conjunction with an MPEG I layer 1 compression scheme. One test compares the performances of the discrete wavelet packet transform ͑DWPT͒ and the modified discrete cosine transform ͑MDCT͒ used in MPEG. Another test compares the performances of a DWPT eight-level nonuniform critical-band split and a DWPT five-level uniform subband split. Results indicate that the critical-band split provides significantly better quality than the uniform subband split for sounds with tonal and strong low-frequency content, while the DWPT outperforms the MDCT with significant improvement for nontonal sounds.

JOURNAL-AUDIO …, 2007

An overview of the recently finalized ISO/MPEG standard for multichannel audio com-pression MPEG Surround is provided. This audio compression scheme enables backward-compatible multichannel audio coding and transmission at unsurpassed coding efficiency. This is ...

2000

Three extension tools for extending and enhancing the com- pression performance of prediction-based lossless audio coding are proposed. The first extension aims at supporting floating- point data input in addition to integer PCM data. The sec- ond is progressive-order prediction of the starting samples at each random-access frame, where the information on previous frame is not available. The third is

Emerging Technologies, 2008. …, 2008

High data quality at low bit rate is an essential goal that people want to achieve. It is necessary to transfer data at low bit rate so that the bandwidth of the medium can be utilized efficiently. In most of the speech coding techniques the goal of low bit rate transfer is achieved but the data quality is affected badly. The proposed technique is an attempt to improve the data quality at low bit rate as well as fast transmission of data. The proposed technique protects the data quality by applying Linear Predictive Coding-10 and achieves the low bit rate by applying Quadrature Mirror Filter. A comprehensive analysis is on the basis of given parameters as size, compression time, Signal to Noise Ratio, power, energy, power in air, energy in air, mean, standard deviation and intensity.

MPEG stands for MOVING PICTURE EXPERTS GROUP is a standard for video and audio compression for eliminating the noisy signals from the transmitted signals from the satellite. Audio compression is a basic method defined under MPEG-1 and MPEG-4 which by coding techniques compress audio signals to filter out undesired signals.This paper focuses on the MPEG technology, need and coding technique for audio compression.

International Journal of Digital Multimedia Broadcasting, 2019

Audio data compression has revolutionised the way in which the music industry and musicians sell and distribute their products. Our previous research presented a novel codec named ACER (Audio Compression Exploiting Repetition), which achieves data reduction by exploiting irrelevancy and redundancy in musical structure whilst generally maintaining acceptable levels of noise and distortion in objective evaluations. However, previous work did not evaluate ACER using subjective listening tests, leaving a gap to demonstrate its applicability under human audio perception tests. In this paper, we present a double-blind listening test that was conducted with a range of listeners (N=100). The aim was to determine the efficacy of the ACER codec, in terms of perceptible noise and spatial distortion artefacts, against de facto standards for audio data compression and an uncompressed reference. Results show that participants reported no perceived differences between the uncompressed, MP3, AAC, A...

As Multimedia Technology is growing day by day over past decades, therefore demand for digital information increasing rapidly. This digital information contains multimedia files like image files, audio files that require a large space so no other option than compression. In Compression high input stream of data is converted into small size. Data Compression for audio purposes is a field of digital signal processing that focuses on reducing bit-rate of audio signals to enhance transmission speed and storage capacity of fast developing multimedia by removing large amount of unnecessary duplicate data. The advantages of the compression technique are reduction in storage space, bandwidth, transmission power and energy. This paper is based on transform technology for compression of the audio signal. In this methodology, different transforms such as Discrete Wavelet Transform (DWT) and Discrete Cosine Transform (DCT) are used. Mean compression ratio is calculated for DCT & DWT. Performance measures like peak signal-to-noise ratio (PSNR), signal-to-noise ratio (SNR) & normalized root mean square error (NRMSE) are calculated and then compared.

IEEE Transactions on Speech and Audio Processing, 2003

A new quality-scalable high-fidelity multichannel audio compression algorithm based on MPEG-2 Advanced Audio Coding (AAC) is presented in this research. The Karhunen-Loève Transform (KLT) is applied to multichannel audio signals in the pre-processing stage to remove inter-channel redundancy. Then, signals in de-correlated channels are compressed by a modified AAC main profile encoder. Finally, a channel transmission control mechanism is used to re-organize the bitstream so that the multichannel audio bitstream has a quality scalable property when it is transmitted over a heterogeneous network. Experimental results show that, compared with AAC, the proposed algorithm achieves a better performance while maintaining a similar computational complexity at the regular bit rate of 64 kbit/sec/ch. When the bitstream is transmitted to narrow-band end users at a lower bit rate, packets of some channels can be dropped, and slightly degraded yet full-channel audio can still be reconstructed in a reasonable fashion without any additional computational cost.

2014

The present paper deals with the subjective evaluation of audio coding technologies using the “Similarity Rating” psychometric method. Compressed audio excerpts are presented to a group of experienced listeners by pairs of stimuli. Each pair represents a different type of distortion, with compression included. These types of distortion correspond to three psychoacoustic attributes, sharpness, roughness, and fluctuation strength. Listener’s task is to evaluate the degree of similarity between the pairs. The final output is the localization of each measured codec in this psychoacoustic space. Multidimensional scaling is the statistic technique, belonging to Multivariate Statistical methods, which allows processing the results obtained from this method.

International Journal of Electrical and Computer Engineering (IJECE), 2021

Digital audio is required to transmit large sizes of audio information through the most common communication systems; in turn this leads to more challenges in both storage and archiving. In this paper, an efficient audio compressive scheme is proposed, it depends on combined transform coding scheme; it is consist of i) bi-orthogonal (tab 9/7) wavelet transform to decompose the audio signal into low & multi high sub-bands, ii) then the produced sub-bands passed through DCT to de-correlate the signal, iii) the product of the combined transform stage is passed through progressive hierarchical quantization, then traditional run-length encoding (RLE), iv) and finally LZW coding to generate the output mate bit stream. The measures Peak signal-to-noise ratio (PSNR) and compression ratio (CR) were used to conduct a comparative analysis for the performance of the whole system. Many audio test samples were utilized to test the performance behavior; the used samples have various sizes and vary in features. The simulation results appear the efficiency of these combined transforms when using LZW within the domain of data compression. The compression results are encouraging and show a remarkable reduction in audio file size with good fidelity.

International Journal of Computer Applications, 2012

Speech compression is one of the leading vicinity of digital signal processing that spotlight on dipping the bit rate of speech signals for transmission and storage devoid of considerable loss of quality. In past decades many speech coding techniques have been proposed for speech analysis. This paper attempts to assess and compare two compression techniques on speech signals. To execute this idea we have chosen two low bit rate and widely used speech analysis methods called VELP and MELP. The performances of both are evaluated by performing objective quality tests including PESQ, IS and CEP. Similar speech files are tested with both coders. The objective assessments show that at low bit rate the MELP shows better performance as compared to VELP.

Revista de Tecnología, 2015

n this paper, audio perceptual compression systems are described, giving special attention to the former one: The MPEG 1, Layer III, in short, the MP3. Other compression technologies are described, especially the technologies evaluated in the present work: OGG Vorbis, WMA (Windows Media Audio) from Microsoft Corporation and AAC (Audio Coding Technologies).