Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.

Abstract: 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.

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract: 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.

Abstract: 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 downmixed 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.

Abstract: Acoustic echo control and noise suppression is an important part of any “handsfree” telecommunication system, such as telephony or audio or video conferencing systems. Bandwidth and computational complexity constraints have prevented that stereo or multi-channel telecommunication systems have been widely applied. The advantages are very low complexity, high robustness, scalability to multi-channel audio without a need for loudspeaker signal distortion, and efficient integration of echo and noise control in the same algorithm.

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E?1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract: 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.

Abstract: 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.