Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: A method includes establishing digital communication between a first user device and a second user device using a first codec. The method also includes selecting, based on an input signal representing an estimated unfiltered available bandwidth for the digital communication satisfying a first filter selection threshold, a first filter of two or more filters, and filtering the input signal using the first filter. The method further includes determining that the filtered input signal satisfies a first channel bandwidth threshold and, in response to determining that the filtered input signal satisfies the channel bandwidth threshold, selecting a second codec different from the first codec for further digital communication between the first user device and the second user device.

Abstract: An electronic device for adjusting a speech output rate (speech rate) of speech output data.

Abstract: A speech transmission compensation apparatus that assists discrimination of speech heard by a user, includes: one or more computers each including a memory and a processor configured to: accept input of a speech signal, detect a specific type of sound in the speech signal, analyze an acoustic characteristic of the specific type of sound in the speech signal and output the acoustic characteristic; accept input of the acoustic characteristic being output by the memory and the processor, generate a vibration signal of a duration corresponding to the acoustic characteristic and output the vibration signal; and accept input of the vibration signal being output by the memory and the processor and provide the user with vibration for the duration on the basis of the vibration signal.

Abstract: An audio encoding/decoding apparatus and method using vector quantized residual error features are disclosed. An audio signal encoding method includes outputting a bitstream of a main codec by encoding an original signal, decoding the bitstream of the main codec, determining a residual error feature vector from a feature vector of a decoded signal and a feature vector of the original signal, and outputting a bitstream of additional information by encoding the residual error feature vector.

Abstract: A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: Disclosed is a voice recognition method and apparatus using artificial intelligence. A voice recognition method using artificial intelligence may include: generating a utterance by receiving a voice command of a user; obtaining a user's intention by analyzing the generated utterance; deriving an urgency level of the user on the basis of the generated utterance and prestored user information; generating a first response in association with the user's intention; obtaining main vocabularies included in the first response; generating a second response by using the main vocabularies and the urgency level of the user; determining a speech rate of the second response on the basis of the urgency level of the user; and outputting the second response according to the speech rate by synthesizing the second response to a voice signal.

Abstract: A method of analyzing time coherence in the noisy signal including the steps of: a) determining a reference signal from the noisy signal by applying treatment (10, 18) to the noisy signal that is suitable for attenuating speech components more strongly than the noise component, in particular by an adaptive recursive predictive algorithm of the LMS type; b) determining (24) a probability of speech being present/absent on the basis of the respective energy levels in the spectral domain of the noisy signal and of the reference signal; and c) deriving (26) a denoised estimate of the speech signal from the noise signal as a function of the probability of the speech being present/absent as determined in this way.

Abstract: Accepting the speech having the noise superimposed thereon and converting it into a signal on a time axis of the speech, an amplitude component of a speech for each predetermined frequency band of the converted signal on the frequency axis is calculated. Calculating a noise reduction coefficient, the noise component is reduced by multiplying the signal on the frequency axis of the original signal by the calculated noise reduction coefficient. By estimating the target value of the remaining noise for each frequency band, a signal on a frequency axis in which a signal corresponding to a frequency band of which target value estimated by the noise target value is larger than the value of the amplitude component of the signal on the frequency axis of which noise component is reduced is corrected to a signal corresponding to the target value is restored, into a signal on a time axis.

Abstract: Prevalence detection is advantageously applied to the result of specific spectral discrimination to adaptively determine prevalent frequencies existing within an audio signal containing speech. Prevalent frequencies in this audio signal so isolated are attenuated in a highly selective manner, thus reducing the masking potential of pervasive resonances and obfuscative energy within the speech itself over low energy language-imparting speech elements.

Abstract: A method of reducing noise in an audio signal, comprising the steps of: using a furrow filter to select spectral components that are narrow in frequency but relatively broad in time; using a bar filter to select spectral components that are broad in frequency but relatively narrow in time; analyzing the relative energy distribution between the output of the furrow and bar filters to determine the optimal proportion of spectral components for the output signal; and reconstructing the audio signal to generate the output signal. A second pair of time-frequency filters may be used to further improve intelligibility of the output signal. The temporal relationship between the furrow filter output and the bar filter output may be monitored so that the fricative components are allowed primarily at boundaries between intervals with no voiced signal present and intervals with voice components. A noise reduction system for an audio signal.

Abstract: An audio signal encoding method and apparatus for efficiently encoding an audio signal in an interval having many birth sinusoids and enabling tracking of sinusoidal signals in the next interval, and a computer readable recording medium having embodied thereon a computer program for executing the audio signal encoding method are provided. According to the method and apparatus, by applying transform coding instead of parametric coding to a frame having many birth sinusoids, the sinusoids are encoded, thereby reducing the number of bits required for the encoding and enabling efficient coding. Also, when transform coding is applied to a frame of a predetermined interval, an inverse transform of the transform coding is applied to the encoded data in order to decode the data, and then sinusoids are extracted from the decoded data, thereby enabling tracking of sinusoids of the next frame.

Abstract: A system and method of processing sound signals are disclosed. In one embodiment, a speech coder applies a first sound signal enhancement process to a first part of a sound signal and applies a second sound signal enhancement process to a second part of the sound signal. The sound signal is then coded using the enhanced first part of the sound signal and the enhanced first part of the sound signal and the enhanced sound part of the sound signal. Examples of the portions of the sound signal that are separately processed include an excitation signal component and a spectral component of the sound signal.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on non-linear values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: Provided are a method, medium and apparatus for enhancing an acoustic signal using an auditory property. An acoustic signal is enhanced by generating a plurality of harmonic signals based on a predetermined acoustic signal frequency, selecting harmonic signals, which exist in an area masked by the predetermined harmonic signal, from among the generated plurality of harmonic signals, and outputting harmonic signals remaining after excluding the selected harmonic signals from the generated plurality of harmonic signals. The enhancement results in a bass signal of good sound quality and having a low distortion ratio, without changing the structure of a micro speaker.

Abstract: A multi-channel signal enhancement system reinforces signal content and improves the signal-to-noise ratio of a multi-channel signal. The system detects, tracks, and reinforces non-stationary periodic signal components of a multi-channel signal. The periodic signal components of the signal may represent vowel sounds or other voiced sounds. The system may detect, track, or attenuate quasi-stationary signal components in the multi-channel signal.

Abstract: Sound quality and speech comprehensibility are to be improved for hearing device wearers when watching television. Provision is made for this purpose to record acoustic signals at a recording site on a data medium. The acoustic signals are recorded simultaneously with a first microphone as a first recording. The data medium is played back on an individual playback device in an individual environment. Here the signal played back from the data medium is re-recorded as a second recording with a second microphone. The two recordings are connected to each other, subtracted in particular, and the result is used to adjust a hearing device program. It is thus possible to take into account individual acoustic environmental conditions in the hearing device program.