Patent classifications
G10L19/12
AUDIO DECODING DEVICE, AUDIO ENCODING DEVICE, AUDIO DECODING METHOD, AUDIO ENCODING METHOD, AUDIO DECODING PROGRAM, AND AUDIO ENCODING PROGRAM
The purpose of the present invention is to reduce distortion a frequency band component encoded with a small number of bits in a time domain and improve quality. An audio decoding device (10) decodes an encoded audio signal and outputs the audio signal. A decoding unit (10a) decodes an encoded sequence containing an encoded audio signal and obtains a decoded signal. A selective temporal envelope shaping unit (10b) shapes a temporal envelope of a decoded signal in the frequency band on the basis of decoding related information concerning decoding of the encoded sequence.
Apparatus and method realizing a fading of an MDCT spectrum to white noise prior to FDNS application
An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal includes a receiving interface for receiving one or more frames comprising information on a plurality of audio signal samples of an audio signal spectrum of the encoded audio signal, and a processor for generating the reconstructed audio signal. The processor is configured to generate the reconstructed audio signal by fading a modified spectrum to a target spectrum, if a current frame is not received by the receiving interface or if the current frame is received by the receiving interface but is corrupted, wherein the modified spectrum includes a plurality of modified signal samples, wherein, for each of the modified signal samples of the modified spectrum, an absolute value of the modified signal sample is equal to an absolute value of one of the audio signal samples of the audio signal spectrum.
Apparatus and method realizing a fading of an MDCT spectrum to white noise prior to FDNS application
An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal includes a receiving interface for receiving one or more frames comprising information on a plurality of audio signal samples of an audio signal spectrum of the encoded audio signal, and a processor for generating the reconstructed audio signal. The processor is configured to generate the reconstructed audio signal by fading a modified spectrum to a target spectrum, if a current frame is not received by the receiving interface or if the current frame is received by the receiving interface but is corrupted, wherein the modified spectrum includes a plurality of modified signal samples, wherein, for each of the modified signal samples of the modified spectrum, an absolute value of the modified signal sample is equal to an absolute value of one of the audio signal samples of the audio signal spectrum.
Time domain spectral bandwidth replication
A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.
Time domain spectral bandwidth replication
A wireless audio system for encoding and decoding an audio signal using spectral bandwidth replication is provided. Bandwidth extension is performed in the time-domain, enabling low-latency audio coding.
ENCODING DEVICE, DECODING DEVICE, AND COMMUNICATION SYSTEM FOR EXTENDING VOICE BAND
A first encoding unit generates a first encoded signal by encoding a component within a first band in a voice signal. A frequency shifting unit shifts the frequency of a component within a second band in the voice signal, the second band having a frequency higher than that of the first band, to the frequency of a component within the first band. A second encoding unit generates a second encoded signal by encoding the component whose frequency has been shifted in the frequency shifting unit. An output unit outputs both the first encoded signal generated in the first encoding unit and the second encoded signal generated in the second encoding unit.
ENCODING DEVICE, DECODING DEVICE, AND COMMUNICATION SYSTEM FOR EXTENDING VOICE BAND
A first encoding unit generates a first encoded signal by encoding a component within a first band in a voice signal. A frequency shifting unit shifts the frequency of a component within a second band in the voice signal, the second band having a frequency higher than that of the first band, to the frequency of a component within the first band. A second encoding unit generates a second encoded signal by encoding the component whose frequency has been shifted in the frequency shifting unit. An output unit outputs both the first encoded signal generated in the first encoding unit and the second encoded signal generated in the second encoding unit.
APPARATUS AND METHOD FOR SELECTING ONE OF A FIRST ENCODING ALGORITHM AND A SECOND ENCODING ALGORITHM USING HARMONICS REDUCTION
An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.
APPARATUS AND METHOD FOR SELECTING ONE OF A FIRST ENCODING ALGORITHM AND A SECOND ENCODING ALGORITHM USING HARMONICS REDUCTION
An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.
Method and apparatus for providing speech coding coefficients using re-sampled coefficients
A method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate.