Class / Patent application number | Description | Number of patent applications / Date published |
704216000 | Correlation function | 7 |
20080235008 | Sound Masking System and Masking Sound Generation Method - In a masking sound generation apparatus, a CPU analyzes a speech utterance speed of a received sound signal. Then, the CPU copies the received sound signal into a plurality of sound signals and performs the following processing on each of the sound signals. Namely, the CPU divides each of the sound signals into frames on the basis of a frame length determined on the basis of the speech utterance speed. Reverse process is performed on each of the frames to replace a waveform of the frame with a reverse waveform, and a windowing process is performed to achieve a smooth connection between the frames. Then, the CPU randomly rearranges the order of the frames and mixes the plurality of sound signals to generate a masking sound signal. | 09-25-2008 |
20080275697 | Audio Processing - An audio processing apparatus for processing two sampled audio signals to detect a temporal position of one of the audio signals with respect to the other. The apparatus detects audio power characteristics of each signal in respect of successive continuous temporal portions of each of the two signals, the portions having identical lengths and each portion including at least two audio samples, and correlates the detected audio power characteristics in respect of the two audio signals to establish a most likely temporal offset between the two audio signals. | 11-06-2008 |
20090037167 | Apparatus and method of encoding and decoding audio signal - In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels have been independently subdivided into blocks if the first and second channels are not correlated with each other, and the first and second channels have been synchronously subdivided into blocks if the first and second channels are correlated with each other and difference coding is used. The embodiment further includes obtaining subdivision information from the audio data frame. The subdivision information includes first information and second information. The first information indicates whether the first and second channels are independently subdivided or synchronously subdivided, and the second information indicates how the subdividing is performed. The first and second channels are decoded based on the obtained subdivision information. | 02-05-2009 |
20090132242 | PORTABLE AUDIO RECORDING AND PLAYBACK SYSTEM - A portable audio recording and playback system is provided. The portable audio recording and playback system can synchronously display the corresponding lyrics or character contents, and it also can selectively record user's voice or the mixed signal of the user's voice and a background music including music and a human voice of singer during playback. Or the volume of the human voice in the background music can be increased or decreased corresponding to that of the music and be outputted or repeatedly played with changing speech speed without the tone changed for achieving the effects of on-the-go vocal and/or music accompaniment, voice recording, and language learning with changing speech speed without the tone changed. | 05-21-2009 |
20090132243 | CONVERSION DEVICE - A plurality of pairs of segments to be weighted/added are selected non-linearly with respect to a time axis of audio data. A speed conversion is achieved by performing the weighting/addition on the selected pairs of segments. The non-linear selection is performed by (a) obtaining all possible pairs of segments constituting the audio data, (b) calculating a degree of similarity pertaining to each possible pair, (c) ranking the all possible pairs of segments according to the degrees of similarity, and (d) overlapping at least one of the all possible pairs of segments that holds the highest degree of similarity. | 05-21-2009 |
704217000 | Autocorrelation | 2 |
20080215317 | Lossless multi-channel audio codec using adaptive segmentation with random access point (RAP) and multiple prediction parameter set (MPPS) capability - A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point. In an exemplary embodiment in which segments within a frame are of the same duration and a power of two of the analysis block duration, the RAP and/or transient constraints set a maximum segment duration to ensure the desired conditions. RAP and MPPS are particularly applicable to improve overall performance for longer frame durations. | 09-04-2008 |
20090204397 | LINEAR PREDICTIVE CODING OF AN AUDIO SIGNAL - An apparatus for linear predictive coding of an audio signal comprises a segmentation processor ( | 08-13-2009 |