摘要:
The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank (101) configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit (102) configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S. The subband processing unit (102) performs a block based nonlinear processing wherein the magnitude of samples of the synthesis subband signal are determined from the magnitude of corresponding samples of the analysis subband signal and a predetermined sample of the analysis subband signal. In addition, the system comprises a synthesis filterbank (103) configured to generate the time stretched and/or frequency transposed signal from the synthesis subband signal.
摘要:
A device is disclosed. The device includes a plurality of ports to receive a plurality of audio streams, an audio content control unit configured to modify playback length of an audio content of at least one of the plurality of audio streams according to an input time interval, an audio decoder and a memory buffer coupled to the audio decoder and the audio content control unit. The memory buffer is used by the audio content control unit to buffer at least one of the plurality of audio streams.
摘要:
An autocorrelation calculation unit 21 calculates an autocorrelation R O (i) from an input signal. A prediction coefficient calculation unit 23 performs linear prediction analysis by using a modified autocorrelation R' O (i) obtained by multiplying a coefficient w O (i) by the autocorrelation R O (i). It is assumed here, for each order i of some orders i at least, that the coefficient w O (i) corresponding to the order i is in a monotonically increasing relationship with an increase in a value that is negatively correlated with a fundamental frequency of the input signal of the current frame or a past frame.
摘要:
An apparatus for normalizing input data of an acoustic model includes a window extractor configured to extract windows of frame data to be input to an acoustic model from frame data of a speech to be recognized, and a normalizer configured to normalize the frame data to be input to the acoustic model in units of the extracted windows.
摘要:
A method of synchronizing playback of audio data sent over a first wireless network from an audio source to a wireless speaker package that is adapted to play the audio data. The method includes comparing a first time period over which audio data was sent over the first wireless network to a second time period over which the audio data was received by the wireless speaker package, and playing the received audio data on the wireless speaker package over a third time period that is related to the comparison of the first and second time periods.
摘要:
Various disclosed implementations involve processing and/or playback of a recording of a conference involving a plurality of conference participants. Some implementations disclosed herein involve receiving audio data corresponding to a recording of at least one conference involving a plurality of conference participants. In some examples, only a portion of the received audio data will be selected as playback audio data. The selection process may involve a topic selection process, a talkspurt filtering process and/or an acoustic feature selection process. Some examples involve receiving an indication of a target playback time duration. Selecting the portion of audio data may involve making a time duration of the playback audio data within a threshold time difference of the target playback time duration.
摘要:
A noise suppressor comprises a first (401) and a second transformer (403) for generating a first and second frequency domain signal from a frequency transform of a first and second microphone signal. A gain unit (405, 407, 409) determines time frequency tile gains in response to a difference measure for magnitude time frequency tile values of the first frequency domain signal and magnitude time frequency tile values of the second frequency domain signal. A scaler (411) generates a third frequency domain signal by scaling time frequency tile values of the first frequency domain signal by the time frequency tile gains; and the resulting signal is converted to the time domain by a third transformer (413). A designator (405, 407, 415) designates time frequency tiles of the first frequency domain signal as speech tiles or noise tiles; and the gain unit (409) determines the gains in response to the designation of the time frequency tiles as speech tiles or noise tiles.