Preface. 1. Introduction. 2. Signal Processing Essentials. 3. Quantization and Entropy Coding. 4. Linear Prediction in Narrowband and Wideband Coding. 5. Psychoacoustic Principles. 6. Time-Frequency Analysis: Filter Banks and Transforms. 7. Transform Coders. 8. Subband Coders. 9. Sinusoidal Coders. 10. Audio Coding Standards and Algorithms. 11. Lossless Audio Coding and Digital Watermarking. 12. Quality Measures for Perceptual Audio Coding. References. Index.

http://jdsp.engineering.asu.edu/books/cover-page-audio+contents.pdf

Audio Signal Processing and Coding

A systematic review of typical biases encountered in modern audio quality listening tests is presented. The following three types of bias are discussed in more detail: bias due to affective judgments, response mapping bias, and interface bias. In addition, a potential bias due to perceptually nonlinear graphic scales is discussed. A number of recommendations aiming to reduce the aforementioned biases are provided, including an in-depth discussion of direct and indirect anchoring techniques.

/pdf/on-some-biases-encountered-in-modern-audio-quality-listening-4pzderleio.pdf

On Some Biases Encountered in Modern Audio Quality Listening Tests-A Review

In recent years audio coding has become a very popular field for research and applications. Especially perceptual audio coding schemes, such as MPEG-1 Layer-3 (MP3) and MPEG-2 Advanced Audio Coding (AAC), are widely used for efficient storage and transmission of music signals. Nevertheless, for professional applications, such as archiving and transmission in studio environments, lossless audio coding schemes are considered more appropriate. Traditionally, the technical approaches used in perceptual and lossless audio coding have been separate worlds. In perceptual audio coding, the use of filter banks, such as the lapped orthogonal transform “Modified Discrete Cosine Transform” (MDCT), has been the approach of choice being used by many state of the art coding schemes. On the other hand, lossless audio coding schemes mostly employ predictive coding of waveforms to remove redundancy. Only few attempts have been made so far to use transform coding for the purpose of lossless audio coding. This work presents a new approach of applying the lifting scheme to lapped transforms used in perceptual audio coding. This allows for an invertible integer-tointeger approximation of the original transform, e.g. the IntMDCT as an integer approximation of the MDCT. The same technique can also be applied to low-delay filter banks. A generalized, multi-dimensional lifting approach and a noise-shaping technique are introduced, allowing to further optimize the accuracy of the approximation to the original transform. Based on these new integer transforms, this work presents new audio coding schemes and applications. The audio coding applications cover lossless audio coding, scalable lossless enhancement of a perceptual audio coder and fine-grain scalable perceptual and lossless audio coding. Finally an approach to data hiding with high data rates in uncompressed audio signals based on integer transforms is described.

/pdf/audio-coding-based-on-integer-transforms-2tfvc3yz97.pdf

Audio Coding based on Integer Transforms

Techniques and tools related to coding and decoding of audio information are described. For example, redundant coded information for decoding a current frame includes signal history information associated with only a portion of a previous frame. As another example, redundant coded information for decoding a coded unit includes parameters for a codebook stage to be used in decoding the current coded unit only if the previous coded unit is not available. As yet another example, coded audio units each include a field indicating whether the coded unit includes main encoded information representing a segment of an audio signal, and whether the coded unit includes redundant coded information for use in decoding main encoded information.

Sub-band voice codec with multi-stage codebooks and redundant coding

Entropy coding and decoding techniques are described, which may be implemented separately or in combination. For example, a video encoder uses two-layer run level coding to reduce bitrate for frequency transform coefficients in a quick and efficient manner, and a video decoder uses corresponding two-layer run level decoding. This two-layer coding/decoding can be generalized to more than two layers of run level coding/decoding. The video encoder and decoder exploit common patterns in run level information to reduce code table size and create opportunities for early termination of decoding. Using zoned Huffman code tables helps limit overall table size while still providing a level of adaptivity in encoding and decoding. Using embedded Huffman code tables allows the encoder and decoder to reuse codes for 8×8, 8×4, 4×8, and 4×4 blocks.

Multi-layer run level encoding and decoding

A method for error concealment in an encoded audio signal, whereby a set of spectral coefficients are divided into at least two sub-bands(14), whereupon said sub-bands undergo reverse transformation (16). A specific prediction is performed (18) for each quasi- time signal of a sub-band in order to obtain an estimated temporal representation for a sub-band of a set of spectral coefficients following on from a real set. Forward transformation (20) of the time signal of each sub-band provides estimated spectral coefficients that can be used (28) instead of defective spectral coefficients of a subsequent set of spectral coefficients in order to conceal transmission errors, for example. Sub-band transformation provides independence from transformation characteristics such as frame length, window type or MDCT algorithm, while at the same time ensuring that spectral processing is maintained for error concealment, whereby the spectral characteristics of audio signals can also be taken into account during said error concealment.

Method and device for error concealment in an encoded audio-signal and method and device for decoding an encoded audio signal

Error Concealment for Compressed Digital Audio

The invention relates to a device and method for entropy encoding and associated decoding. The inventive code comprises a code table with reversible code words (12) and an escape region for information words to be coded which are located outside the region (14) defined by said code table. Said region can be selected in such a way that a major part of the information words is coded with symmetrical code words by means of the code table. On the one hand, it is possible to carry out forward decoding and also backward decoding (24) and on the other hand, use of reversible code words allows for rapid recognition of errors in a code word stream transmitted over a non-ideal channel.

Device and method for entropy encoding of information words and device and method for decoding entropy-encoded information words

In a method for coding an audio signal to obtain a coded bit stream, discrete-time samples of the audio signal are transformed into the frequency domain to obtain spectral values. The spectral values are coded with a code table having a limited number of code words of different lengths to obtain spectral values coded by code words, the length of a code word assigned to a spectral value being that much shorter the higher the probability of occurrence of the spectral value is. A raster is then specified for the coded bit stream, the raster having equidistant raster points and the distance between the raster points depending on the code table(s) used. In order to obtain error-tolerant Huffman coding, priority code words, which represent particular spectral values which are psycho-acoustically more important than other spectral values, are so arranged in the raster that the start of each priority code word coincides with a raster point.

Method and devices for coding or decoding an audio signal or bit stream

An apparatus (100) for generating spectral replacement values for an audio signal is provided. The apparatus (100) comprises a buffer unit (110) for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus (100) comprises a concealment frame generator (120) for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame comprises filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator (120) is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value.

Apparatus and method for error concealment in low-delay unified speech and audio coding (usac)

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