Abstract: An apparatus for encoding a first channel and a second channel of an audio input signal including two or more channels to obtain an encoded audio signal according to an embodiment includes a normalizer configured to determine a normalization value for the audio input signal depending on the first channel of the audio input signal and depending on the second channel of the audio input signal. Moreover, the apparatus includes an encoding unit configured to generate a processed audio signal having a first channel and a second channel. The encoding unit is configured to encode the processed audio signal to obtain the encoded audio signal.

Abstract: A method of encoding an audio signal is provided comprising: applying multiple different time-frequency transformations to an audio signal frame; computing measures of coding efficiency across multiple frequency bands for multiple time-frequency resolutions; selecting a combination of time-frequency resolutions to represent the frame at each of the multiple frequency bands based at least in part upon the computed measures of coding efficiency; determining a window size and a corresponding transform size; determining a modification transformation; windowing the frame using the determined window size; transforming the windowed frame using the determined transform size; modifying a time-frequency resolution within a frequency band of the transform of the windowed frame using the determined modification transformation.

Abstract: Methods and apparatus for performing variable block length watermarking of media are disclosed. Example apparatus include means for evaluating a masking ability of a first audio block; means selecting a first frequency to represent a first code, the means for selecting to (i) select the first frequency selected from a first set of frequencies that are detectable when performing a frequency transformation using a first block length, but are not detectable when performing a frequency transformation using a second block length, and (ii) select a second frequency to represent a second code, the second frequency selected from a second set of frequencies that are detectable when performing a frequency transformation using the second block length; means for synthesizing a first signal having the first frequency with the masking ability of the first audio block; and means for combining the first signal with the first audio block.

Abstract: Decoding of a first message is disclosed, wherein first and second messages are encoded by a code (represented by a state machine) to produce first and second code words, which are received over a communication channel. A plurality of differences (each corresponding to a hypothesized value of a part of the first message) between the first and second messages are hypothesized. An initial code word segment is selected having, as associated previous states, a plurality of initial states (each associated with a hypothesized difference and uniquely defined by the hypothesized value of the part of the first message).

Abstract: Decoding of a first message is disclosed, wherein first and second messages are encoded by a code (represented by a state machine) to produce first and second code words, which are received over a communication channel. A plurality of differences (each corresponding to a hypothesized value of a part of the first message) between the first and second messages are hypothesized. An initial code word segment is selected having, as associated previous states, a plurality of initial states (each associated with a hypothesized difference and uniquely defined by the hypothesized value of the part of the first message).

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: The present invention relates to a speech processing apparatus, a speech processing method and a program which, when multichannel audio signals are downmixed and coded, prevent delay and an increase in the computation amount upon decoding of the audio signals. An inverse multiplexing unit (101) acquires coded data on which a BC parameter is multiplexed. An uncorrelated frequency-time transform unit (102) performs IMDCT transform and IMDST transform of frequency spectrum coefficients of a monaural signal (XM) obtained from this coded data to generate the monaural signal XM) which is a time domain signal and a signal (XD?) which is substantially uncorrelated with this monaural signal (XM). The stereo synthesis unit (103) generates a stereo signal by synthesizing the monaural signal (XM) and the signal (XD?) using the BC parameter. The present invention is applicable to, for example, a speech processing apparatus which decodes a downmixed and coded stereo signal.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: The present invention discloses a method and a system for converting vocal sounds into digital data format. This technique will significantly decrease the amount of memory needed to store the digital data of the recorded voice. The system is comprised of a microphone converting the vocal sound signals into electrical signal, amplifying and filtering module for analyzing the electrical signals, a comparator module for comparing the analog signal to pre-defined value and sampling by clock edge module for representing the output signal of the comparator as a digital data format, a memory module for storing said digital data, a filtering module for reducing the alternating the analog signal higher harmonics, an amplifying module increasing the filtered signals amplitude and transducer module for converting the electrical amplifying signals into vocal sound signal.