Digital Audio Compression

IEEE Transactions on Speech and Audio Processing, 1995

A new audio transform coding technique is proposed that reduces the bitrate requirements of the Perceptual Transform Audio Coders, by utilizing the stationarity characteristics of the audio signals. The method detects the frames which have significant audible content and codes them in a way similar to conventional Perceptual Transform Coders. However, when successive data frames are found to be similar to those sections, then their audible differences are only coded. An error analysis for the proposed method is presented and results from tests on different types of audio material are listed, indicating that an average of 30% in compression gain (over the conventional Perceptual Audio Coders bitrate) can be achieved, with a small deterioration in the audio quality of the coded signal. The proposed method has the advantage of easy adaptation within the Perceptual Transform Coders architecture and add only small computational overhead to these systems. n recent years the introduction of digital audio as a method for storing, processing and transmitting high-fidelity acoustic signals has helped in the evolution of numerous applications in the field of consumer electronics and professional audio. It is also envisaged that, in the near future, significant new techniques will become commercially available and novel applications will emerge based on the manipulation of audio data within multimedia or audiovisual technologies. However, the feasibility of such future applications, as well as some current ones, greatly depends on the use of data compression techniques which reduce the data transmission rate and memory storage requirements. Given the existence of such techniques, terrestrial or satellite transmission channels can be economically employed for single or multi-channel audio data transmission, and also data storage media can be efficiently utilized for storing lengthy segments of acoustic signals. The storage and transmission of such high-quality audio data (here it will be considered as reference the Compact Disc format, based on a 44.1 kHz sampling rate and 16-bit resolution) results in the relatively high bit-rate of 706 kBits/s, per data channel. This data rate can be technically or economically prohibitive for many applications, and this necessitates the introduction of data compression, preferably by using low-complexity methods (so that real-time implementations are not impeded), and without the insertion of perceptually detectable distortions. Applications which have emerged or are expected to appear with strong dependence on such signal compression technology , are in the area of high-fidelity audio for radio broadcasting (especially for the Digital Audio Broadcasting -DAB format [2]), in the area of multichannel audio for HDTV, in storing and processing of audio signals for domestic (e.g. multimedia or home studio) and professional applications (multichannel music recording), in transmitting audio data through computer or communication networks (e.g. ISDN), etc. Coding and data compression methods for acoustic signals have been known for at least 4 decades, but until recently they were mainly concerned with speech signals [3], [4]. More recently, Transform Manuscript

[1991] Proceedings, Advanced Computer Technology, Reliable Systems and Applications, 1991

Based on frequency-domain techniques, coding of high-quality audio with bit rates down to 64 kbit/s is possible. This performance is achieved using perceptual coding. Transform coding can be used to get the best performance at very low bit rates. Real-time implementations of several types of low bit rate codecs have been developed. Standardization of low bit rate audio coding systems

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.

2010

This paper presents a theory of lossless digital compression. Quality of voice signal is not important for voice communication. In hearing music high quality music is always recommended. For this emphasis is given on the quality of speech signal. To save more music it is needed to save them consuming smaller memory space. In proposed compression 8-bit PCM/PCM speech signal is compressed. When values of samples are varying they are kept same. When they are not varying the number of samples containing same value is saved. After compression the signal is also an 8bit PCM/PCM. MPEG-4 ALS is applied in this compressed PCM signal for better compression.

Indonesian Journal of Electrical Engineering and Computer Science, 2018

In recent years, multichannel audio systems are widely used in modern sound devices as it can provide more realistic and engaging experience to the listener. This paper focuses on the performance evaluation of three lossy, i.e. AAC, Ogg Vorbis, and Opus, and three lossless compression, i.e. FLAC, TrueAudio, and WavPack, for multichannel audio signals, including stereo, 5.1 and 7.1 channels. Experiments were conducted on the same three audio files but with different channel configurations. The performance of each encoder was evaluated based on its encoding time (averaged over 100 times), data reduction, and audio quality. Usually, there is always a trade-off between the three metrics. To simplify the evaluation, a new integrated performance metric was proposed that combines all the three performance metrics. Using the new measure, FLAC was found to be the best lossless compression, while Ogg Vorbis and Opus were found to be the best for lossy compression depends on the channel configuration. This result could be used in determining the proper audio format for multichannel audio systems.

IEEE Transactions on Consumer Electronics, 1999

Scientific Journal of Informatics, 2019

Audio file size is relatively larger when compared to files with text format. Large files can cause various obstacles in the form of large space requirements for storage and a long enough time in the shipping process. File compression is one solution that can be done to overcome the problem of large file sizes. Arithmetic coding is one algorithm that can be used to compress audio files. The arithmetic coding algorithm encodes the audio file and changes one row of input symbols with a floating point number and obtains the output of the encoding in the form of a number of values greater than 0 and smaller than 1. The process of compression and decompression of audio files in this study is done against several wave files. Wave files are standard audio file formats developed by Microsoft and IBM that are stored using PCM (Pulse Code Modulation) coding. The wave file compression ratio obtained in this study was 16.12 percent with an average compression process time of 45.89 seconds, whil...

This chapter presents an introduction to speech compression techniques, together with a detailed description of speech/audio compression standards including narrowband, wideband and fullband codecs. We will start with the fundamental concepts of speech signal digitisation, speech signal characteristics such as voiced speech and unvoiced speech and speech signal representation. We will then discuss three key speech compression techniques, namely waveform compression, parametric compression and hybrid compression methods. This is followed by a consideration of the concept of narrowband, wideband and fullband speech/audio compression. Key features of standards for narrowband, wideband and fullband codecs are then summarised.

Proceedings. IEEE SoutheastCon 2001 (Cat. No.01CH37208), 2001

With the spread of the Internet into mainstream society, there has come a demand for the efficient transmission of multimedia information. Accompanying the drive to find more efficient ways of utilizing limited transmission bandwidth is a need to find novel ways of compressing data. This thesis proposed the utilization of transform coding compression techniques for the transmission of audio data across networks. The Discrete Cosine Transform (DCT) and the Discrete Sine Transform (DST) were the primary transforms utilized. This thesis investigated the viability of utilizing individual transforms, as well as, nested modifications of these transforms for compression purposes. These techniques were compared to those already in existence. Viability was determined using objective compression measures. It was found that transform coding techniques gave a useful alternative to the techniques in existence. A voice-over-IP (VOIP) application that utilized one of the transform coding techniques was implemented. v

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.