AnOverview of the Coding StandardMPEG-4 Audio Amendments 1 and 2 : HE-AAC , SSC , and HE-AAC v 2
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paper_title: High-frequency regeneration in speech coding systems

paper_content:

The traditional method of high-frequency regeneration (HFR) of the excitation signal in baseband coders has been to rectify the transmitted baseband, followed by spectral flattening. In addition, a noise source is added at high frequencies to compensate for lack of energy during certain sounds. In this paper, we reexamine the whole HFR process. We show that the degree of rectification does not affect the output speech, and that, with proper processing, the high-frequency noise source may be eliminated. We introduce a new type of HFR based on spectral duplication of the baseband. Two types of spectral duplication are presented: spectral folding and spectral translation. Finally, in order to eliminate the problem of breaking the harmonic structure due to spectral duplication, we propose a pitch-adaptive spectral duplication scheme in the frequency domain by using adaptive transform coding to code the baseband.
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paper_title: A new technique for wideband enhancement of coded narrowband speech

paper_content:

Telephone speech is typically bandlimited to 4 kHz, resulting in a 'muffled' quality. Coding speech with a bandwidth greater than 4 kHz reduces this distortion, but requires a higher bit rate to avoid other types of distortion. An alternative to coding wider bandwidth speech is to exploit correlations between the 0-4 kHz and 4-8 kHz speech bands to re-synthesize wideband speech from decoded narrowband speech. This paper proposes a new technique for highband spectral envelope prediction, based upon codebook mapping with codebooks split by voicing. An objective comparison with several existing methods reveals that this new technique produces the smallest highband spectral distortion. Combined with a suitable highband excitation synthesis scheme, this envelope prediction scheme produces a significant quality improvement in speech that has been coded using narrowband standards.
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paper_title: BANDWIDTH EXTENSION OF AUDIO SIGNALS BY SPECTRAL BAND REPLICATION

paper_content:

Spectral Band Replication (SBR) is a new audio coding tool that significantly improves the coding gain of perceptual coders and speech coders. Currently, there are three different audio coders that have shown a vast improvement by the combination with SBR: MPEG-AAC, MPEG-Layer II and MPEG-Layer III (mp3), all three being parts of the open ISO-MPEG standard. The combination of AAC and SBR will be used in the standardized Digital Radio Mondiale (DRM) system, and SBR is currently also being standardized within MPEG-4. SBR is a so-called bandwidth extension technique, where a major part of a signal’s bandwidth is reconstructed from the lowband on the receiving side. It is developed and marketed by Coding Technologies, an international company in the audio coding field. This paper will focus on the technical details of SBR and in particular on the filter bank, which is the basis of the SBR process.
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paper_title: BANDWIDTH EXTENSION OF AUDIO SIGNALS BY SPECTRAL BAND REPLICATION

paper_content:

Spectral Band Replication (SBR) is a new audio coding tool that significantly improves the coding gain of perceptual coders and speech coders. Currently, there are three different audio coders that have shown a vast improvement by the combination with SBR: MPEG-AAC, MPEG-Layer II and MPEG-Layer III (mp3), all three being parts of the open ISO-MPEG standard. The combination of AAC and SBR will be used in the standardized Digital Radio Mondiale (DRM) system, and SBR is currently also being standardized within MPEG-4. SBR is a so-called bandwidth extension technique, where a major part of a signal’s bandwidth is reconstructed from the lowband on the receiving side. It is developed and marketed by Coding Technologies, an international company in the audio coding field. This paper will focus on the technical details of SBR and in particular on the filter bank, which is the basis of the SBR process.
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paper_title: Speech coding: a tutorial review

paper_content:

The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for cellular communications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. We feel that this approach will not only point out key references but will also provide valuable background to the beginner. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. We then proceed with descriptions of waveform coders, sinusoidal transform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. Finally, we present concluding remarks followed by a discussion of opportunities for future research. >
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paper_title: Perceptual coding of digital audio

paper_content:

During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby AC-2/AC-3, and the Sony ATRAC/SDDS algorithms. Then, we describe subjective evaluation methodologies in some detail, including the ITU-R BS.1116 recommendation on subjective measurements of small impairments. This paper concludes with a discussion of future research directions.
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paper_title: Adaptive Signal Models: Theory, Algorithms, and Audio Applications

paper_content:

List of Figures. Foreword M. Vetterli. Preface. 1. Signal Models and Analysis-Synthesis. 2. Sinusoidal Modeling. 3. Multiresolution Sinusoidal Modeling. 4. Residual Modeling. 5. Pitch-Synchronous Models. 6. Matching Pursuit and Atomic Models. 7. Conclusions. Appendix A: Two-Channel Filter Banks. Appendix B: Fourier Series Representations. References. References for Poetry Excerpts. Index.
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paper_title: A tone oriented voice excited vocoder

paper_content:

An LPC base-band vocoder is developed. The novel feature concerns the coding of the base-band. A model is set up for the base-band as a set of modulated tones. Algorithms are presented for the extraction of amplitude and phase/frequency of the tones. Implementation aspects as well as simulation results are discussed. Total bit rates in the order of 3,2-4.8 kbits are possible where approximately one half of the bits represents the base-band coding. Experiments have shown the coder to be robust to background noise.
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paper_title: Nonstationary spectral modeling of voiced speech

paper_content:

The main purpose of this paper is to present a novel model for voiced speech. The classical model, which is being used in many applications, assumes local stationarity, and consequently imposes a simple and well known line structure to the short-time spectrum of voiced speech. The model derived in this paper allows for local non-stationarities not only in terms of pitch perturbations, but in terms of vocal tract variations as well. The resulting structure of the short-time spectrum becomes more complex, but can still be interpreted in terms of generalized lines. The proposed model supports new forms of spectral prediction, which can be put to advantage in speech coding applications. Experimental results are presented supporting the validity of both the model itself and the prediction relationships. Finally, a new class of speech coders, denoted harmonic coders, based on the presented model, is proposed, and a specific implementation is presented.
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paper_title: Design of a scalable parametric audio coder

paper_content:

Parametric coding (or compression) of audio signals is a promising new technique that has found its way into recent coding standards like MPEG-4, demonstrating its ability to achieve high audio quality at very low bit-rates. While transform coding techniques (like MP3 and AAC) rely on a perception model, parametric coding techniques complement the perception model with a signal model. The components of the parametric signal model are the following.
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paper_title: Speech analysis/Synthesis based on a sinusoidal representation

paper_content:

A sinusoidal model for the speech waveform is used to develop a new analysis/synthesis technique that is characterized by the amplitudes, frequencies, and phases of the component sine waves. These parameters are estimated from the short-time Fourier transform using a simple peak-picking algorithm. Rapid changes in the highly resolved spectral components are tracked using the concept of "birth" and "death" of the underlying sine waves. For a given frequency track a cubic function is used to unwrap and interpolate the phase such that the phase track is maximally smooth. This phase function is applied to a sine-wave generator, which is amplitude modulated and added to the other sine waves to give the final speech output. The resulting synthetic waveform preserves the general waveform shape and is essentially perceptually indistinguishable from the original speech. Furthermore, in the presence of noise the perceptual characteristics of the speech as well as the noise are maintained. In addition, it was found that the representation was sufficiently general that high-quality reproduction was obtained for a larger class of inputs including: two overlapping, superposed speech waveforms; music waveforms; speech in musical backgrounds; and certain marine biologic sounds. Finally, the analysis/synthesis system forms the basis for new approaches to the problems of speech transformations including time-scale and pitch-scale modification, and midrate speech coding [8], [9].
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paper_title: Towards flexible audio coding

paper_content:

thesis is about audio coding and improving flexibility thereof. Audio coding is used to reduce the bit rate needed to represent audio signals in digital format. When the available bit rate is low, parametric coding methods that use models to describe perceptually-important features of audio signals have shown their efficiency. We introduce several improvements to sinusoidal coding, a parametric coding method that is based on sinusoidal modeling of audio. Flexibility is important since new applications bring a need for coders that can operate over a large range of possible bit rates and are able to represent different types of audio material. Flexibility can be obtained by combining into one coder a set of subcoders that can efficiently represent different types or features of audio signals, and properly complement each other at different rates. For true flexibility, methods that allow fast coder adaptation should be deployed. We develop methods for rate-distortion optimal real-time design of quantizers in audio coding. This thesis consists of seven research papers. In paper A, we introduce a signal pre-processing method that facilitates removal of pre-echo artifacts when coding signals containing transients (sharp attacks). Papers B-F are devoted to sinusoidal audio coding. In papers B and C, we present improvements to the matching-pursuit sinusoidal estimation method. In papers D, E, and F, we consider quantization of sinusoidal parameters. We apply high-rate quantization theory to find the asymptotically optimal rate distribution between sinusoids and the corresponding asymptotically optimal quantizers for sinusoidal parameters, such that a perceptual distortion measure is minimized under a given rate constraint. The quantizers are derived analytically, which allows a coder to adapt quickly to changing bit-rate requirements. Paper G is devoted to multistage audio coding, a coding method where subcoders are combined in a cascaded way. We use high-rate theory to develop a flexible analytical framework for the asymptotically optimal rate distribution between subcoders and the design of the corresponding asymptotically optimal quantizers.
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paper_title: HILN-the MPEG-4 parametric audio coding tools

paper_content:

The MPEG-4 Audio Standard combines tools for efficient and flexible coding of audio. For very low bitrate applications, tools based on a parametric signal representation are utilised. The parametric speech coding tools (HVXC) are already available in Version 1 of MPEG-4. The main focus of this paper is on the parametric audio coding tools "Harmonic and Individual Lines plus Noise" (KILN) which are included in Version 2 of MPEG-4. As already indicated by their name, the HILN tools are based on the decomposition of the audio signal into components which are described by appropriate source models and represented by model parameters. This paper gives an overview of the HILN tools, presents the recent advances in signal modelling and parameter coding, and concludes with an evaluation of the subjective audio quality.
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paper_title: ADVANCES IN PARAMETRIC CODING FOR HIGH-QUALITY AUDIO

paper_content:

Compared to traditional waveform coding standards that employ subband or transform coding, parametric coding is regarded as a technique that allows even further reduction of bit rates. In response to a call for proposals by MPEG-4 in January 2001, Philips submitted a parametric coder targetting at a high quality level. Currently, this coder is improved in the course of the “MPEG-4 Extension 2” standardisation process. Recently, two enhancements have been made increasing the overall performance of the coder. These consist of an improved model for the noise object and a parametric stereo extension.
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paper_title: A new speech coding model based on a least-squares sinusoidal representation

paper_content:

In recent years the concept of analysis-by-synthesis has been applied very successfully to improving the performance of LPC based models. At the same time, new speech models have been introduced based on representing speech by a sum of amplitude and frequency-modulated sinusoids which have been shown to successfully represent the non-linear, time-varying and quasi-periodic nature of speech. In this paper we present an approach to applying the analysis-by-synthesis technique to sinusoidal speech modelling in an attempt to increase the ability of the model to accurately represent the speech waveform.
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paper_title: ADVANCES IN PARAMETRIC CODING FOR HIGH-QUALITY AUDIO

paper_content:

Compared to traditional waveform coding standards that employ subband or transform coding, parametric coding is regarded as a technique that allows even further reduction of bit rates. In response to a call for proposals by MPEG-4 in January 2001, Philips submitted a parametric coder targetting at a high quality level. Currently, this coder is improved in the course of the “MPEG-4 Extension 2” standardisation process. Recently, two enhancements have been made increasing the overall performance of the coder. These consist of an improved model for the noise object and a parametric stereo extension.
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paper_title: Autoregressive Modeling of Temporal Envelopes

paper_content:

Autoregressive (AR) models are commonly obtained from the linear autocorrelation of a discrete-time signal to obtain an all-pole estimate of the signal's power spectrum. We are concerned with the dual, frequency-domain problem. We derive the relationship between the discrete-frequency linear autocorrelation of a spectrum and the temporal envelope of a signal. In particular, we focus on the real spectrum obtained by a type-I odd-length discrete cosine transform (DCT-Io) which leads to the all-pole envelope of the corresponding symmetric squared Hilbert temporal envelope. A compact linear algebra notation for the familiar concepts of AR modeling clearly reveals the dual symmetries between modeling in time and frequency domains. By using AR models in both domains in cascade, we can jointly estimate the temporal and spectral envelopes of a signal. We model the temporal envelope of the residual of regular AR modeling to efficiently capture signal structure in the most appropriate domain.
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paper_title: Bark and ERB Bilinear Transforms

paper_content:

Use of a bilinear conformal map to achieve a frequency warping nearly identical to that of the Bark frequency scale is described. Because the map takes the unit circle to itself, its form is that of the transfer function of a first-order allpass filter. Since it is a first-order map, it preserves the model order of rational systems, making it a valuable frequency warping technique for use in audio filter design. A closed-form weighted-equation-error method is derived that computes the optimal mapping coefficient as a function of sampling rate, and the solution is shown to be generally indistinguishable from the optimal least-squares solution. The optimal Chebyshev mapping is also found to be essentially identical to the optimal least-squares solution. The expression 0.8517[arctan(0.06583fs)]/sup 1/2/-0.916 is shown to accurately approximate the optimal allpass coefficient as a function of sampling rate f/sub s/ in kHz for sampling rates greater than 1 kHz. A filter design example is included that illustrates improvements due to carrying out the design over a Bark scale. Corresponding results are also given and compared for approximating the related "equivalent rectangular bandwidth (ERB) scale" of Moore and Glasberg (ACTA Acustica, vo.82, p.335-45, 1996) using a first-order allpass transformation. Due to the higher frequency resolution called for by the ERB scale, particularly at low frequencies, the first-order conformal map is less able to follow the desired mapping, and the error is two to three times greater than the Bark-scale case, depending on the sampling rate.
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paper_title: Matching pursuit with damped sinusoids

paper_content:

The matching pursuit algorithm derives an expansion of a signal in terms of the elements of a large dictionary of time-frequency atoms. This paper considers the use of matching pursuit for computing signal expansions in terms of damped sinusoids. First, expansion based on complex damped sinusoids is explored; it is shown that the expansion can be efficiently derived using the FFT and simple recursive filterbanks. Then, the approach is extended to provide decompositions in terms of real damped sinusoids. This extension relies on generalizing the matching pursuit algorithm to derive expansions with respect to dictionary subspaces; of specific interest is the subspace spanned by a complex atom and its conjugate. Developing this particular case leads to a framework for deriving real-valued expansions of real signals using complex atoms. Applications of the damped sinusoidal decomposition include system identification, spectral estimation, and signal modeling for coding and analysis-modification-synthesis.
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paper_title: Sum-difference stereo transform coding

paper_content:

An audio filterbank coder using both monophonic and stereophonic redundancy and irrelevancy is described. This coder, with a transform filterbank, uses both signal-processing gains and perceptual gains to provide transparent or nearly transparent coding of CD and R-DAT audio signals at 192 kb/s/stereo pair, and high-quality coding at 128 kb/s/stereo pair. Methods of using the interchannel redundancies and psychoacoustic characteristics are described. >
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paper_title: Spatial Audio Processing: MPEG Surround and Other Applications

paper_content:

This book collects a wealth of information about spatial audio coding into one comprehensible volume. It is a thorough reference to the 3GPP and MPEG Parametric Stereo standards and the MPEG Surround multi-channel audio coding standard. It describes key developments in coding techniques, which is an important factor in the optimization of advanced entertainment, communications and signal processing applications. Until recently, technologies for coding audio signals, such as redundancy reduction and sophisticated source and receiver models did not incorporate spatial characteristics of source and receiving ends. Spatial audio coding achieves much higher compression ratios than conventional coders. It does this by representing multi-channel audio signals as a downmix signal plus side information that describes the perceptually-relevant spatial information. Written by experts in spatial audio coding, Spatial Audio Processing: reviews psychoacoustics (the relationship between physical measures of sound and the corresponding percepts) and spatial audio sound formats and reproduction systems; brings together the processing, acquisition, mixing, playback, and perception of spatial audio, with the latest coding techniques; analyses algorithms for the efficient manipulation of multiple, discrete and combined spatial audio channels, including both MP3 and MPEG Surround; shows how the same insights on source and receiver models can also be applied for manipulation of audio signals, such as the synthesis of virtual auditory scenes employing head-related transfer function (HRTF) processing and stereo to N-channel audio upmix. Audio processing research engineers and audio coding research and implementation engineers will find this an insightful guide. Academic audio and psychoacoustic researchers, including post-graduate and third/fourth year students taking courses in signal processing, audio and speech processing, and telecommunications, will also benefit from the information inside.
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paper_title: Parametric Coding of Stereo Audio

paper_content:

Parametric-stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus a small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psychoacoustical principles. The monaural signal can be encoded using any (conventional) audio coder. Experiments show that the parameterized description of spatial properties enables a highly efficient, high-quality stereo audio representation.
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paper_title: Spatial Audio Processing: MPEG Surround and Other Applications

paper_content:

This book collects a wealth of information about spatial audio coding into one comprehensible volume. It is a thorough reference to the 3GPP and MPEG Parametric Stereo standards and the MPEG Surround multi-channel audio coding standard. It describes key developments in coding techniques, which is an important factor in the optimization of advanced entertainment, communications and signal processing applications. Until recently, technologies for coding audio signals, such as redundancy reduction and sophisticated source and receiver models did not incorporate spatial characteristics of source and receiving ends. Spatial audio coding achieves much higher compression ratios than conventional coders. It does this by representing multi-channel audio signals as a downmix signal plus side information that describes the perceptually-relevant spatial information. Written by experts in spatial audio coding, Spatial Audio Processing: reviews psychoacoustics (the relationship between physical measures of sound and the corresponding percepts) and spatial audio sound formats and reproduction systems; brings together the processing, acquisition, mixing, playback, and perception of spatial audio, with the latest coding techniques; analyses algorithms for the efficient manipulation of multiple, discrete and combined spatial audio channels, including both MP3 and MPEG Surround; shows how the same insights on source and receiver models can also be applied for manipulation of audio signals, such as the synthesis of virtual auditory scenes employing head-related transfer function (HRTF) processing and stereo to N-channel audio upmix. Audio processing research engineers and audio coding research and implementation engineers will find this an insightful guide. Academic audio and psychoacoustic researchers, including post-graduate and third/fourth year students taking courses in signal processing, audio and speech processing, and telecommunications, will also benefit from the information inside.
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paper_title: Parametric Coding of Stereo Audio

paper_content:

Parametric-stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus a small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psychoacoustical principles. The monaural signal can be encoded using any (conventional) audio coder. Experiments show that the parameterized description of spatial properties enables a highly efficient, high-quality stereo audio representation.
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paper_title: LOW COMPLEXITY PARAMETRIC STEREO CODING IN MPEG-4

paper_content:

Parametric stereo coding in combination with a State-of-the-Art coder for the underlying monaural audio signal results in the most ef cient coding scheme for stereo signals at very low bit rates available today. This paper reviews those aspects of the parametric stereo paradigm that are important for audio coding applications. A complete parametric stereo coding system is presented, which was recently standardized in MPEG-4 Audio. Using complex modulated lter banks, it allows implementation with low computational complexity. The system is backward compatible and enables high quality stereo coding at total bit rate of 24 kbit/s when used in combination with aacPlus.
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paper_title: Spatial Audio Processing: MPEG Surround and Other Applications

paper_content:

This book collects a wealth of information about spatial audio coding into one comprehensible volume. It is a thorough reference to the 3GPP and MPEG Parametric Stereo standards and the MPEG Surround multi-channel audio coding standard. It describes key developments in coding techniques, which is an important factor in the optimization of advanced entertainment, communications and signal processing applications. Until recently, technologies for coding audio signals, such as redundancy reduction and sophisticated source and receiver models did not incorporate spatial characteristics of source and receiving ends. Spatial audio coding achieves much higher compression ratios than conventional coders. It does this by representing multi-channel audio signals as a downmix signal plus side information that describes the perceptually-relevant spatial information. Written by experts in spatial audio coding, Spatial Audio Processing: reviews psychoacoustics (the relationship between physical measures of sound and the corresponding percepts) and spatial audio sound formats and reproduction systems; brings together the processing, acquisition, mixing, playback, and perception of spatial audio, with the latest coding techniques; analyses algorithms for the efficient manipulation of multiple, discrete and combined spatial audio channels, including both MP3 and MPEG Surround; shows how the same insights on source and receiver models can also be applied for manipulation of audio signals, such as the synthesis of virtual auditory scenes employing head-related transfer function (HRTF) processing and stereo to N-channel audio upmix. Audio processing research engineers and audio coding research and implementation engineers will find this an insightful guide. Academic audio and psychoacoustic researchers, including post-graduate and third/fourth year students taking courses in signal processing, audio and speech processing, and telecommunications, will also benefit from the information inside.
---
paper_title: LOW COMPLEXITY PARAMETRIC STEREO CODING IN MPEG-4

paper_content:

Parametric stereo coding in combination with a State-of-the-Art coder for the underlying monaural audio signal results in the most ef cient coding scheme for stereo signals at very low bit rates available today. This paper reviews those aspects of the parametric stereo paradigm that are important for audio coding applications. A complete parametric stereo coding system is presented, which was recently standardized in MPEG-4 Audio. Using complex modulated lter banks, it allows implementation with low computational complexity. The system is backward compatible and enables high quality stereo coding at total bit rate of 24 kbit/s when used in combination with aacPlus.
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Title: An Overview of the Coding Standard MPEG-4 Audio Amendments 1 and 2: HE-AAC, SSC, and HE-AAC v2

Section 1: Introduction
Description 1: Introduce the historical context and motivation behind MPEG-4 Audio Amendments 1 and 2, detailing the focus areas and objectives of the amendments.

Section 2: MPEG-4 SBR
Description 2: Discuss the Spectral Band Replication (SBR) technology, its basic principles, and its significance in audio coding.

Section 3: SBR Encoding Process
Description 3: Provide an overview of the SBR encoding process, including the various stages and the role of complex-valued filter banks.

Section 4: SBR Bit Stream
Description 4: Describe the structure and characteristics of the SBR bit stream and how it ensures consistent coding of transients.

Section 5: SBR Decoding Process
Description 5: Explain the steps involved in the SBR decoding process, highlighting the role of QMF analysis and synthesis.

Section 6: Other Aspects
Description 6: Discuss additional aspects of SBR, such as low-power SBR, downsampled SBR, and scalable systems, focusing on their implementation and applications.

Section 7: MPEG-4 SSC
Description 7: Detail the principles and components of the Sinusoidal Coder (SSC), including transient, sinusoidal, and noise modeling.

Section 8: SSC Decoder
Description 8: Outline the SSC decoding process, describing the operations of the transient, sinusoidal, and noise synthesizers.

Section 9: SSC Encoder
Description 9: Explain the SSC encoding process, including the detection and analysis of transients, sinusoids, and noise.

Section 10: SSC Bit Stream
Description 10: Describe the structure and content of the SSC bit stream, including quantization and encoding methods.

Section 11: Parametric Stereo Coding
Description 11: Discuss the Parametric Stereo (PS) tool, its development, and application alongside SSC and HE-AAC for effective stereo image coding.

Section 12: Stereo Analysis
Description 12: Explain the process of stereo analysis within the PS encoder, detailing the extraction of spatial parameters and down-mixing techniques.

Section 13: Stereo Synthesis
Description 13: Describe the process of stereo synthesis in the PS decoder, focusing on the generation of stereo output signals from mono-downmix and PS parameters.

Section 14: SSC Performance
Description 14: Provide performance results and subjective evaluation of SSC, comparing it with existing technologies and discussing the results of listening tests.

Section 15: MPEG-4 HE-AAC v2
Description 15: Introduce the combination of HE-AAC with the PS tool, explaining the resulting increase in coding efficiency for stereo signals at low bit rates.

Section 16: Combining HE-AAC with PS
Description 16: Detail the integration of the PS tool with HE-AAC, highlighting the computational efficiency and flexibility of the HE-AAC v2 decoder.

Section 17: Conclusions
Description 17: Summarize the advancements and improvements brought by MPEG-4 Audio Amendments 1 and 2, emphasizing their impact on low bit rate applications and overall audio quality.