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Oliver Kunz

Martin Dietz

Karlheinz Brandenburg

Bodo Teichmann

Uwe Gbur

Heinz Gerhauser

Heinz Gerh&auml;user

J&uuml;rgen Herre

Rainer Buchta

Martin Sieler

Juergen Herre

Ralph Sperschneider

Charles Quito Robinson

Grant Allen Davidson

J&uuml;rgen Zeller

Michael Beer

Ali Nowbakht-Irani

Klaus Peichl

Marina Bosi

Jurgen Zeller

Jurgen Herre

In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

Process of low sampling rate digital encoding of audio signals

In a device for processing a stereo audio signal having a first channel and a second channel the stereo signal is at first analyzed to obtain a measure for a quantity of bits required by a coder to code the stereo audio signal using a coding algorithm. The first channel and the second channel are then modified when the measure for the quantity of bits is larger than a predetermined value, the modification being performed in such a way that the energy of a sum signal of the first and the second modified channel is in a predetermined relation to the energy of a sum signal of the first and the second channel and that a difference signal of the first and the second modified channel is attenuated in contrast to the difference signal of the first and the second channel. Especially for audio coders requiring a constant output bit rate the side channel is attenuated in the case of stereo audio signals, the coding of which cannot meet the output bit rate of the coder, by which a stereo channel separation is abandoned for the benefit of an increased audio bandwidth or a reduction of quantizing disturbances, respectively.

Method and device for processing a stereo audio signal

In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.

Method for signalling a noise substitution during audio signal coding

A method and a device for entropy encoding and associated decoding make use of a code consisting on the one hand of a code table with reversible code words ( ) and comprising on the other hand an escape region for information words to be coded which are located outside the region ( ) 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 the code table. On the one hand, it is thus possible to carry out, in addition to forward decoding, also backward decoding ( ) 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

A method for coding or decoding an audio signal combines the advantages of TNS processing and noise substitution. A time-discrete audio signal is initially transformed to the frequency domain in order to obtain spectral values of the temporal audio signal. Subsequently, a prediction of the spectral values in relation to frequency is carried out in order to obtain spectral residual values. Within the spectral residual values, areas are detected encompassing spectral residual values with noise properties. The spectral residual values in the noise areas are noise-substituted, whereupon information concerning the noise areas and noise substitution is incorporated into side information pertaining to a coded audio signal. Thus, considerable bit savings in case of transient signals can be achieved.

Method for coding an audio signal

A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.

Method subband of coding and decoding audio signals using variable

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