Patent application number | Description | Published |
20100061558 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR PROVIDING A SET OF SPATIAL CUES ON THE BASIS OF A MICROPHONE SIGNAL AND APPARATUS FOR PROVIDING A TWO-CHANNEL AUDIO SIGNAL AND A SET OF SPATIAL CUES - An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels. | 03-11-2010 |
20100061566 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving a signal, by an audio processing apparatus; computing a long-term power and a short-term power by estimating power of the signal; generating a slow gain based on the long-term power; generating a fast gain based on the short-term power; obtaining a final gain by combining the slow gain and the fast gain; and, modifying the signal using the final gain. | 03-11-2010 |
20100085117 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving, by an audio processing apparatus, a signal, and feedback information estimated based on a normalizing gain; generating a noise estimation based on the signal; computing a gain filter for noise canceling, based on the noise estimation and the signal; and, obtaining a restricted gain filter by applying the feedback information to the gain filter. | 04-08-2010 |
20100296669 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; receiving user gain input; generating a linear gain factor and a non-linear gain factor using the user gain input; modifying the non-linear gain factor using absolute threshold of hearing and power of the input signal to generate a modified non-linear gain factor; and, applying the linear gain factor and the modified non-linear gain factor to the audio signal. | 11-25-2010 |
20100310085 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, an input signal; estimating indicator function using a signal power of the input signal; obtaining an adapted filter using the indicator function and an equalization filter; and, generating an output signal by applying the adapted filter to the input signal. | 12-09-2010 |
20110286609 | MULTIPLE MICROPHONE BASED DIRECTIONAL SOUND FILTER - A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module. | 11-24-2011 |
20110299702 | APPARATUS, METHOD AND COMPUTER PROGRAM FOR PROVIDING A SET OF SPATIAL CUES ON THE BASIS OF A MICROPHONE SIGNAL AND APPARATUS FOR PROVIDING A TWO-CHANNEL AUDIO SIGNAL AND A SET OF SPATIAL CUES - An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels. | 12-08-2011 |
20130230184 | ECHO SUPPRESSION COMPRISING MODELING OF LATE REVERBERATION COMPONENTS - An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters. | 09-05-2013 |
Patent application number | Description | Published |
20080201153 | Generation of Multi-Channel Audio Signals - A decoder ( | 08-21-2008 |
20080267413 | Method to Generate Multi-Channel Audio Signal from Stereo Signals - A perceptually motivated spatial decomposition for two-channel stereo audio signals, capturing the information about the virtual sound stage, is proposed. The spatial decomposition allows to re-synthesize audio signals for playback over other sound systems than two-channel stereo. With the use of more front loudspeakers, the width of the virtual sound stage can be increased beyond +/−30° and the sweet spot region is extended. Optionally, lateral independent sound components can be played back separately over loudspeakers on the two sides of a listener to increase listener envelopment. It is also explained how the spatial decomposition can be used with surround sound and wavefield synthesis based audio system. According to the main embodiment of the invention applying to multiple audio signals, it is proposed to generate multiple output audio signals (y | 10-30-2008 |
20090067634 | Enhancing Audio With Remixing Capability - One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability. | 03-12-2009 |
20100241438 | METHOD AND AN APPARATUS OF DECODING AN AUDIO SIGNAL - A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting a ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced. | 09-23-2010 |
20100250259 | METHOD AND AN APPARATUS OF DECODING AN AUDIO SIGNAL - The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal. And, a signal output unit for outputting an ambient component signal is arranged to have an output direction different from that of another signal output unit for outputting a source component signal, whereby a listener can be provided with an audio signal of which ambient sound is enhanced. | 09-30-2010 |
Patent application number | Description | Published |
20100106270 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method of processing an audio signal is disclosed. The present invention comprises receiving downmix signal including object signals, transforming the downmix signal per frequency band, determining a direction of an object signal from the transformed downmix signal, and determining blind information by estimating a level of the object signal corresponding to the direction. Accordingly, the present invention generates blind information in case of using an encoder incapable of generating object information, thereby enabling a gain and panning of object to be controlled using the blind information. | 04-29-2010 |
20100189266 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - A method of processing an audio signal is disclosed. The present invention comprises receiving a downmix signal, object information and preset information, generating downmix processing information using the object information and the preset information, processing the downmix signal using the downmix processing information, and generating multi-channel information using the object information and the preset information, wherein the preset information is extracted from a bitstream. Accordingly, a gain and panning of an object can be easily controlled without user's setting for each object using preset information set in advance. And, a gain and panning of an object can be controlled using preset information modified based on a selection made by a user. | 07-29-2010 |
Patent application number | Description | Published |
20090150161 | SYNCHRONIZING PARAMETRIC CODING OF SPATIAL AUDIO WITH EXTERNALLY PROVIDED DOWNMIX - Embodiments of the present invention are directed to a binaural cue coding (BCC) scheme in which an externally provided audio signal (e.g., a studio engineering audio signal) is transmitted, along with derived cue codes, to a receiver instead of an automatically downmixcd audio signal. The cue codes are (adaptively) synchronized with the externally provided audio signal to compensate for time lags (and changes in those time lags) between the externally downmixed audio signal and the multi-channel signal used to generate the cue codes. If the receiver is a legacy receiver, then the studio engineered audio signal will typically provide a higher-quality playback than would be provided by the automatically downmixed audio signal. If the receiver is a BCC-capable receiver, then the synchronization of the cue codes with the externally provided audio signal will typically improve the quality of the synthesized playback. | 06-11-2009 |
20090319281 | CUE-BASED AUDIO CODING/DECODING - Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels. | 12-24-2009 |
20090319282 | DIFFUSE SOUND SHAPING FOR BCC SCHEMES AND THE LIKE - In one embodiment, C input audio channels are encoded to generate E transmitted audio channel(s), where one or more cue codes are generated for two or more of the C input channels, and the C input channels are downmixed to generate the E transmitted channel(s), where C>E≧1. One or more of the C input channels and the E transmitted channel(s) are analyzed to generate a flag indicating whether or not a decoder of the E transmitted channel(s) should perform envelope shaping during decoding of the E transmitted channel(s). In one implementation, envelope shaping adjusts a temporal envelope of a decoded channel generated by the decoder to substantially match a temporal envelope of a corresponding transmitted channel. | 12-24-2009 |
20110164756 | Cue-Based Audio Coding/Decoding - Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “.1” indicates a single low-frequency effects (LFE) channel and “.2” indicates two LFE channels. | 07-07-2011 |
20120314879 | PARAMETRIC JOINT-CODING OF AUDIO SOURCES - The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme. | 12-13-2012 |