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This article explores the integration of model-based and data-driven approaches within the realm of neural speech and audio coding systems. It highlights the challenges posed by the subjective evaluation processes of speech and audio codecs and discusses the limitations of purely data-driven approaches, which often require inefficiently large architectures to match the performance of model-based methods. The study presents hybrid systems as a viable solution, offering significant improvements to the performance of conventional codecs through meticulously chosen design enhancements. Specifically, it introduces a neural network-based signal enhancer that is designed to postprocess existing codecs’ output, along with the autoencoder-based end-to-end models and LPCNet–hybrid systems that combine linear predictive coding (LPC) with neural networks. Furthermore, the article delves into predictive models that operate within custom feature spaces (TF-Codec) or predefined transform domains (MDCTNet) and examines the use of psychoacoustically calibrated loss functions to train end-to-end neural audio codecs. Through these investigations, the article demonstrates the potential of hybrid systems to advance the field of speech and audio coding by bridging the gap between traditional model-based approaches and modern data-driven techniques.

For decades, linear predictive (LP) analysis of the real-valued time-domain signals has been developed for speech coding, analysis, and synthesis. Recently, the complex-valued frequency-domain LP coding was developed to enhance the estimation performance of the temporal envelope. To apply it for audio coding, a suitable representation for the complex-valued frequency-domain LP coefficients (CLPC) is required before quantization, but there was no efficient way to represent it. To address the problem, we propose efficient CLPC representations that retain some useful properties of conventional LPC representations. Through quantitative and qualitative evaluations, we demonstrate that our proposed representations increase quantization efficiency and improve audio coding performance.

Delivering high-quality spatial audio in the Ambisonics format requires extensive data bandwidth, which may render it inaccessible for many low-bandwidth applications. Existing widely-available multi-channel audio compression codecs are not designed to consider the characteristic inter-channel relations inherent to the Ambisonics format, and thus may not leverage this knowledge to optimise the compression. Therefore, this article proposes a spatial audio compression algorithm, based on a novel reformulation of the Higher-Order Directional Audio Coding (HO-DirAC) method, which is specifically intended for compressing higher-order Ambisonic audio streams. The methodology builds upon the concept of a spherical filter bank acting in the spherical harmonic domain. This results in directionally constrained sound-field estimates and parameterization, which may be utilized to reconstruct the input Ambisonic signals with minimal perceived loss of quality. The results of a listening experiment indicate high perceptual quality when using six or more audio transport channels to deliver fifth-order (36 channels) Ambisonic sound scenes. The proposed formulation is also designed with low computational complexity in mind and may therefore be well suited for compressing Ambisonic sound scenes for a wide range of applications.

2022 AUG 13 (NewsRx) -- By a News Reporter-Staff News Editor at Obesity, Fitness & Wellness Week -- Data detailed on Life Science have been presented. According to news originating [...]

In 1988 the International Standardization Organization (ISO) established the Motion Picture Expert Group (MPEG) to develop a digital coding standard for video and audio signals in order to enable interactive video and audio signals on digital storage media. The MPEG Audio Group started with members from 14 research institutions in order to develop a digital audio coding standard guided by a chairman. As a result the MPEG-1, Layer I, Layer H and Layer III coding standards have been developed and proposed for coding of stereo audio signals at 2 x 192 kbit/s, 2 x 128 kbit/s and 2 x 64 kbit/s in 1992. Later, the abbreviation 'mp3' or 'MP3' was introduced in order to substitute the long name of the successful MPEG-1, Layer III coding standard. This paper describes the development of the MP3 coding standard and its essential components as contributed by the members of the MPEG Audio Group. Index Terms--Audio coding, MPEG standards, MP3

The paper presents a new low complexity design of Psycho-Acoustic Model (PAM), which is the key technology for MPEG-2/4 Advanced Audio Coding (AAC) encoding. The real-time constraint of MPEG AAC leads to a heavy computational bottleneck on today's portable devices. To overcome this problem, design analysis and optimization of PAM are addressed at the algorithmic level. First, the dominant calculation of spreading junction is replaced with small look-up tables. Second, Modified-Discrete-Cosine-Transform-based (MDCT-based) PAM with block type decision in frequency domain is proposed to reduce the overall AAC encoder complexity and maintain the quality. This technique is different from the traditional methods. The computational complexity of PAM is reduced by 82% in total. The proposed design could be implemented in a real-time MPEG-2/4 AAC stereo encoder at Low Complexity profile and bitrate128 kb/s with computational complexity of less than 20 MOPS while maintaining CD quality (1). Index Terms--Audio encoder, Low complexity, MPEG Advanced Audio Coding, Psycho-acoustic Model.

Intellectual property solutions provider Via Licensing Corp. reported the upcoming availability of Extended High Efficiency AAC (xHE-AAC) as part of the Advanced Audio Coding (AAC) patent pool. The expanded patent [...]

Entropy coding, which is an essential part of audio compression, is always required to manage the tradeoffs between compression efficiency and computational complexity, and the strategy to achieve them highly depends on the distributions of inputs. In this paper, we present a method of controlling them for enhancing the compression efficiency of Golomb-Rice (GR) encoding, one of the simplest entropy coding methods optimal for Laplacian distributions. We will show that the proposed invertible and low-complexity mapping of integers enables the GR encoding to assign nearly the optimal code length for a wider range of distributions, generalized Gaussian distributions, maintaining low computational cost. A simulation by random numbers reveals that the proposed coder based on this scheme works about 6 times faster than the state-of-the-art arithmetic coder for Gaussian-distributed integers maintaining the increase in relative redundancy around 2.6%, which is much lower than that of a conventional GR coder. Additionally, an application to a practical speech and audio coding scheme is presented, and an objective evaluation for real speech and audio signals confirms the advantages of the proposed method in compression. The method is expected to widen the capability of low-complexity entropy coding, providing us with more flexible codec designs.

9775945240 High-fidelity multichannel audio coding, 2d ed. Yang, Dai Tracy et al. Hindawi Publishing Corporation 2005 228 pages $119.95 Hardcover EURASIP book series on signal processing and communications; v.1 TK7881 [...]

9775945089 High-fidelity multichannel audio coding. Yang, Dai Tracy et al. Hindawi Publishing Corporation 2004 252 pages $119.95 Hardcover EURASIP book series on signal processing and communications; v.1 TK7881 The authors, [...]

This paper presents a new implementation of the solid state audio player equipped with TwinVQ decoder. TwinVQ is an advanced high quality coding technology, and its basic algorithm is utilized in MPEG-4/Audio standardized by ISO (International Standardization Organization). Using TwinVQ coding, 44.1 kHz sampled audio data can be compressed at a very low bitrate of 40 kbit/s/ch with better sound quality than conventional coding methods as MPEG-1/Audio layer-3. Another feature of the player is that it is extremely compact, business-card-sized, 10 mm thick, including battery, as a result of the efficient design. Decoding algorithms are optimized and implemented into 16 bit fixed-point DSP, and a very thin rechargeable lithium polymer battery is contained integral with the player. As a result, the new player has very practical performance with low power consumption, and 6 hours of 44 kHz stereo audio can be continuously played when the battery is fully charged.

The MPEG-1 audio standard (ISO/IEC 11172-3) establishes guidelines for the compression of high-quality digital audio signals [1]. The standard dictates the function of an encoder/decoder pair (codec), leaving form intentionally vague to allow for competing implementations. A typical approach to real-time operation is to design an application-specific integrated circuit (ASIC) dedicated to encoding, decoding [2], or both. We present an alternative codec that makes use of the general-purpose digital signal processing (DSP) chips that are now common in multimedia-capable workstations and personal computers. We discuss how selective optimization of codec structure allows robust performance using limited resources, highlight some of the problems inherent in translating the abstractions of the standard into assembly code, and point towards further investigations of real-time implementations of communications standards.

A new three-dimensional (3D) audio coding approach is presented to improve the spatial perceptual quality of 3D audio. Different from other audio coding approaches, the distance side information is also quantified, and the non-uniform perceptual quantization is proposed based on the spatial perception features of the human auditory system, which is named as concentric spheres spatial quantization (CSSQ) method. Comparison results were presented, which showed that a better distance perceptual quality of 3D audio can be enhanced by 5.7%∼8.8% through extracting and coding the distance side information comparing with the directional audio coding, and the bit rate of our coding method is decreased of 8.07% comparing with the spatial squeeze surround audio coding.

CLEVELAND -- 'Hello caller, you're on the air.' It's a greeting heard on thousands of radio stations across this country every day. However, without the engineering skill and determination of [...]

Unified speech and audio coding (USAC) is one of the latest coding technologies. It is based on a switchable coding structure, and has demonstrated the highest levels of performance for both speech and music contents. In this paper, we propose an extended version of USAC with a single-mode of operation—which does not require a switching system—by extending the linear prediction-coding mode. The main concept of this extension is the adoption of the advantages of frequency-domain coding schemes, such as windowing and transition control. Subjective test results indicate that the proposed scheme covers speech, music, and mixed streams with adequate levels of performance. The obtained quality levels are comparable with those of USAC.

There is a considerable performance gap between the current scalable audio coding schemes and a nonscalable coder operating at the same bitrate. This suboptimality results from the independent coding of the layers in these systems. One of the aspects that plays a role in this suboptimality is the entropy coding. In practical audio coding systems including MPEG advanced audio coding (AAC), the transform domain coefficients are quantized using an entropy-constrained quantizer. In MPEG-4 scalable AAC (S-AAC), the quantization and coding are performed separately at each layer. In case of Huffman coding, the redundancy introduced by the entropy coding at each layer is larger at lower quantization resolutions. Also, the redundancy for the overall coder becomes larger as the number of layers increases. In fact, there is a tradeoff between the overall redundancy and the fine-grain scalability in which the bitrate per layer is smaller and more layers are required. In this paper, a fine-grain scalable coder for audio signals is proposed where the entropy coding of a quantizer is made scalable via joint design of entropy coding and quantization. By constructing a Huffman-like coding tree where the internal nodes can be mapped to the reconstruction points, the tree can be pruned at any internal node to control the rate-distortion (RD) performance of the encoder in a fine-grain manner. A set of metrics and a trellis-based approach is proposed to create a coding tree so that an appropriate path is generated on the RD plane. The results show the proposed method outperforms the scalable audio coding performed based on reconstruction error quantization as used in practical systems, e.g., in S-AAC.

Byline: JEFFREY C. RIEDMILLER Last month's article discussed the theory behind high-end audio coding. The second half of this series will define broadcast quality and tandem coding losses, and look [...]

Vimicro International Corporation, a multimedia semiconductor and IP-based surveillance solution provider, announced that on Aug. 25, five key government agencies formed an alliance to promote the adoption of the Surveillance [...]

Byline: Jack Kontney One company that has managed to remain relatively obscure despite making gargantuan contributions to the pro audio field is Fraunhofer IIS. Headquartered in Germany, the Fraunhofer Society [...]

In the rapidly emerging market for information goods, a key strategic issue is control of content. For example, it is rare for a week to go by without some news of another technological advance, legal development, or business realignment that has its roots in the development of the MP3 open standard for audio compression/decompression. It has been convincingly argued that the Internet, in particular the MP3 standard, is changing the music industry. The development of the MP3 open standard for audio-video compression, the ready availability of free software for MP3 encoding and playback of digital music, and the ease with which MP3 files are transported across the Internet combine to create a large and growing threat to the recording industry. This threat is manifested in both legitimate uses of MP3 compression for electronic-distribution of music, and what is commonly referred to as Internet piracy, involving violation of music copyrights.

Byline: JEFFREY C. RIEDMILLER Today many broadcasters and cable programmers are under increasing pressure both to prepare and to distribute their content to viewers over more diverse pathways. These next-generation [...]

Fraunhofer has developed MPEG Surround real-time encoding and decoding at bit rates as low as 48 kbit/s for high-quality 5.1 surround sound. Another example for the work of Fraunhofer in [...]

TK7881 977-5945-13-5 High-fidelity multichannel audio coding. Yang, Dai Tracy et al. (EURASIP book series on signal processing and communications; v.1) Hindawi Publishing Corporation, ©2004 252 p. $79.95 (pa) The authors, [...]

Digital audio coding delays have become increasingly critical in real-time wireless applications. In live productions, a codec with ultra low delay is required within the constraints of the available channel bandwidth. However, such a threshold can hardly be reached by means of standard audio coding schemes. To achieve low delay as well as to satisfy cost and power consumption constraints, this paper presents an ultra low delay audio coder by very short block processing and embedded coding implemented in fixed-point DSP. The short block two dimensional (2D) spatial-frequency processing of audio input signal fully exploits the correlation for better compression performance. Lifting wavelet transform with boundary effects minimized by changing wavelet shape is developed using bit shifts and additions to replace multiplications in a fixed-point specification under accuracy constraint. The embedded coding offers the error resilience feature so that joint source-channel coding scheme for unequal error protection can be easily designed by varying both source coding bit rate and channel coding redundancy. Experimental results demonstrate that the proposed coder is efficient and requires less memory in fixed-point computation which guarantees no overflow.

Focuses on MP3 technology and its implications for record companies. Function of MP3s; Software versatility and control as the appeal for MP3s; Reponse to dropping sales attributed to MP3s by record companies; Other technologies proposed, such as the Secure Digital Music Initiative, by record companies to make money from the MP3s; Software by Microsoft to reduce free downloading of music on the Internet.

This paper describes a new audio coding scheme based on adaptive wavelet analysis that provides transparent audio coding for CD-audio signals at low bit rates (≈1.4 bits/sample per channel). A new perceptual cost function is defined to obtain the best wavelet-packet base for each audio frame. The sharp variations in quantization noise that appear at the border of the frames are minimized by a novel approach that avoids overlapping. The proposed coder guarantees high perceptual quality using filters that generate wavelets of any compact support, because a bit-allocation algorithm that takes into account the equivalent filter frequency responses of the synthesis filter bank branches is used.

Chinas DRA audio coding standard is shown to be a barebone transform coder, utilizing transient-localized MDCT for improved pre-echo suppression and statistic allocation of codebooks for high entropy coding efficiency. A signal path of up to 24 bits is provided throughout the codec so that highest audio quality can be delivered if bit rate suffices. Results of five ITU-R BS.1116 compliant subjective listening tests are presented.

Psychoacoustical models have been used extensively within audio coding applications over the past decades. Recently, parametric coding techniques have been applied to general audio and this has created the need for a psychoacoustical model that is specifically suited for sinusoidal modelling of audio signals. In this paper, we present a new perceptual model that predicts masked thresholds for sinusoidal distortions. The model relies on signal detection theory and incorporates more recent insights about spectral and temporal integration in auditory masking. As a consequence, the model is able to predict the distortion detectability. In fact, the distortion detectability defines a (perceptually relevant) norm on the underlying signal space which is beneficial for optimisation algorithms such as rate-distortion optimisation or linear predictive coding. We evaluate the merits of the model by combining it with a sinusoidal extraction method and compare the results with those obtained with the ISO MPEG-1 Layer I-II recommended model. Listening tests show a clear preference for the new model. More specifically, the model presented here leads to a reduction of more than 20% in terms of number of sinusoids needed to represent signals at a given quality level.

A perceptual audio coder, in which each audio segment is adaptively analyzed using either a sinusoidal or an optimum wavelet basis according to the time-varying characteristics of the audio signals, has been constructed. The basis optimization is achieved by a novel switched filter bank scheme, which switches between a uniform filter bank structure (discrete cosine transform) and a non-uniform filter bank structure (discrete wavelet transform). A major artifact of the International ISO/Moving Pictures Experts Group (MPEG) audio coding standard (MPEG-I layers 1 and 2) known as pre-echo distortion which uses a uniform filter bank structure for audio signal analysis, is almost eliminated in the proposed coder. A perceptual masking model implemented using a high-resolution wavelet packet filter bank with 27 subbands, closely mimicking the critical bands of the human auditory system, is employed in this audio coder. The resulting scheme is a variable bit-rate audio coder, which provides compression ratios comparable to MPEG-I layers 1 and 2 with almost transparent quality.

This paper presents complete optimized architecture of Advanced Audio Coder quantization with Huffman coding. After that psychoacoustic model theory is presented and few algorithms described: standard Two Loop Search, its modifications, Genetic, Just Noticeable Level Difference, Trellis-Based and its modification: Cascaded Trellis-Based Algorithm.

This paper proposes a new quantitative method called the transform coefficients, which is based on the algebraic quantization method. This coefficient is defined as a unique amplitude multiplied by a number of pulses.In order to determine the signs of pulse, location and amplitude, we selected the transmission coefficient by adopting an ideal standard errors. This simple quantification has been implemented in scalable coding of microwave transform, and has been proved that its voice signal and music effect is better than that of the SPIHT.

The article discusses why storing MP3 files in Microsoft Corp.'s OneDrive cloud storage system and playing the music through he company's Groove Music mobile app is the best method for storing and retrieving such files. It cites the inability of competitors like Spotify and Amazon to play locally saved MP3 files as the advantage of Groove Music.

The future MPEG‐4 standard will adopt an object‐oriented encoding strategy whereby an audio source is encoded at a very low bit‐rate by adapting a suitable coding scheme to the local characteristics of the signal. One of the most delicate issues in this approach is that the overall performance of the audio encoder greatly depends on the accuracy with which the input signal is classified. This paper shows that the difficult problem of audio classification for object‐oriented coding can be effectively solved by selecting a salient set of acoustic parameters and adopting a fuzzy model for each audio object, obtained by a soft computing‐hybrid learning tool. The audio classifier proposed operates at two levels: recognition of the class to which the input signal belongs (talkspurt, music, noise, signaling tones) and then recognition of the subclass to which it belongs. The results obtained show that fuzzy logic is a valid alternative to the matching techniques of a traditional pattern recognition approach.

In this study the subjective performance of four low‐complexity audio data compression methods are compared, operating at nominal bit rates of 2, 3, 4, and 5 bits per sample, applied to four 20‐kHz bandwidth, 16‐bits per sample digitized musical signals. The simple compression schemes compared were elementary differential pulse‐code modulation (DPCM), noise feedback coding DPCM (NFC‐DPCM), adaptive quantizer DPCM (DPCM‐AQB), and a recently proposed method known as recursively indexed quantizer DPCM (RIQ‐DPCM). Pairs consisting of a reconstructed signal and a reference signal were presented in a two‐interval preference experiment. The reference signals were processed for specified levels of modulated noise reference unit (MNRU) in order to estimate the equality threshold rating (ETR) of the reconstructed audio stimuli. The subjective MNRU values were found to increase by 2–5 dB for each increment in bits per sample. The DPCM‐AQB scores were found to be 8–10 dB higher than for DPCM and NFC‐DPCM. RIQ‐DPCM was rated highest, exceeding the DPCM‐AQB results by 2–5 dB in all tests. Objective measurements of segmental signal‐to‐noise ratio (SNRSEG) for the reconstructed signals predicted a performance level 2–5 dB lower than was actually found in the subjective results, particularly for SNRSEG values below 25 dB.

The search for a flexible and concise alternate representation for digital musical sound leads to the proposal for the use of the MIDI (Musical Instrument Digital Interface) protocol. The problem becomes one of automating the conversion process from sound to MIDI. This requires processing musical sound and extracting the information necessary to represent the sound as MIDI data. We have conducted studies which have led to algorithms for segmentation of the sound and pitch detection of the individual notes. We describe a novel method for pitch detection using subset selection with dictionaries containing harmonic spectra from samples of musical sounds. Examples demonstrating applicability to monophonic sounds as well as signals with multiple sound sources are given, including detection of objects in a complex background scene.

The article focuses on the uses of audio podcasts for library users, recommends podcasts listening apps and discusses which kinds of features are most useful for apps. These include devices that plays MP3 file audio format such as desktop computers and smartphones types of podcast programming such as interviews and storytelling and media companies who are into podcasting including internet media company BuzzFeed Inc. and newspaper company New York Times Co.

Part II. Compares three computer software used for compressing digital audio data. Windows Media Audio from Microsoft; MP3 from Fraunhofer; Advanced Audio Coding.

The article takes a look at the history of portable music technology, beginning with the 8-track player. A time-line is established from 1965 to 2007, charting the volume of sales for each music format. The four music players highlighted are the Dynamite 8, The Sony Walkman, the Discman, and the iPhone.

Discusses Shawn Fanning who wrote the source code for the music file-sharing program, Napster. Details of the program which allows its users to swap music files with one another directly; Suing of Napster by the Recording Industry Association of America (RIAA) which claims that the program facilitates theft of intellectual property; Development of ISO-MPEG Audio Layer-3, or MP3, which compresses CD-quality sound files.

Examines a number of products to use when creating a home network. Products that personalize the delivery of audio and video without personal computers; Services that connect television to the Internet; Ways to transport MP3s to stereo systems; Home gateways that cable and DSL providers will offer that combine networking with a high-speed modem. INSET: THE LIST.