Abstract: The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

Abstract: Disclosed is a system and computer program product of encoding audio content and corresponding method. The method includes determining a characteristic of the audio content, the characteristic of the audio content including at least one of a type or a property of the audio content. Also the method includes classifying the audio content based on the characteristic of the audio content and determining probabilities for multiple predefined audio coding symbols associated with the audio content by calculating a probability for each of the audio coding symbols based on the result of the classification, the probability for an audio coding symbol indicating a frequency at which the audio coding symbol occurs in the audio content. Further, the method encoded the audio content based on the audio coding symbols and the corresponding probabilities to obtain a code value, the code value representing a compression coding format of the audio content.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: Methods and corresponding apparatuses for transmitting and receiving audio signals are described. A transformation is performed on the audio signals in units of frame in order to obtain transformed audio data of each frame, said transformed audio data consisting of multiple signal components in the frequency domain. These signal components of each frame are distributed into multiple adjacent packets in order to generate packets in which signal components distributed from multiple frames are interleaved. Subsequently, the generated packets are transmitted. Accordingly, in case that packet loss occurs during transmission, the audio signals can be recovered based on the received signal components without consuming additional bandwidth. Therefore, robustness against packet loss can be achieved with little overhead.

Abstract: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

Abstract: Embodiments of the present invention relate to video content assisted audio object extraction. A method of audio object extraction from channel-based audio content is disclosed. The method comprises extracting at least one video object from video content associated with the channel-based audio content, and determining information about the at least one video object. The method further comprises extracting from the channel-based audio content an audio object to be rendered as an upmixed audio signal based on the determined information. Corresponding system and computer program product are also disclosed.

Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that results from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets (412, 413) is described. A lost packet (412) is a packet which is deemed to be lost by a transform-based audio decoder. Each of the one or more lost packets (412, 413) comprises a set of transform coefficients (313). A set of transform coefficients (313) is used by the transform-based audio decoder to generate a corresponding frame (412, 413) of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet (412) of the one or more lost packets (412, 413) a number of preceding lost packets from the one or more lost packets (313); wherein the determined number is referred to as a loss position.

Abstract: Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.

Abstract: In a conference call having a plurality of participants interacting in a conference exchange of information in a digital transmission environment, the interaction being across a variable network transmission resource, a method of allocating the level of transmission resource, the methods including the steps of: (a) monitoring predetermined aspects of the participant's behavior during the conference call; (b) determining a divergence of participants behavior from normative values; (c) utilizing any divergence as an indicator of aberrant operation of the participants; and (d) allocating the resource determinative on the divergence of participants behavior from normative values.

Abstract: Example embodiments disclosed herein relate to spatial congruency adjustment. A method for adjusting spatial congruency in a video conference is disclosed. The method in unwarping a visual scene captured by a video endpoint device into at least one rectilinear scene, the video endpoint device being configured to capture the visual scene in an omnidirectional manner, detecting spatial congruency between the at least one rectilinear scene and an auditory scene captured by an audio endpoint device that is positioned in relation to the video endpoint device. The spatial congruency being a degree of alignment between the auditory scene and the at least one rectilinear scene and in response to the detected spatial congruency being below the threshold, adjusting the spatial congruency. Corresponding system and computer program products are also disclosed.

Abstract: An audio spatial rendering apparatus and method are disclosed. In one embodiment, The audio spatial rendering apparatus includes a rendering unit for spatially rendering an audio stream so that the reproduced far-end sound is perceived by a listener as originating from at least one virtual spatial position, a real position obtaining unit for obtaining a real spatial position of a real sound source, a comparator for comparing the real spatial position with the at least one virtual spatial position; and an adjusting unit for, where the real spatial position is within a predetermined range around at least one virtual spatial position, or vice versa, adjusting the parameters of the rendering unit so that the at least one virtual spatial position is changed.

Abstract: The disclosure relates to handling nuisance in teleconference system. An endpoint device (400) for use in a teleconference includes an acquiring unit (401), a judging unit (402), a controller (403) and a processing unit (404). The acquiring unit acquires a media stream for presentation in the teleconference, and receives information from another device. The information includes a first estimation on whether the media stream is a nuisance to the teleconference. As the nuisance to a teleconference, audio or video signals are perceived by users as actually not relevant to the conference session or causing unpleasant feeling or confusion. The judging unit decides whether the media stream is the nuisance at least based on the information. The controller controls the processing of the media stream to degrade or suppress the presentation of the media stream in case that the media stream is decided as the nuisance. The processing unit processes the media stream under the control of the controller.

Abstract: Example embodiments disclosed herein relate to spatial congruency adjustment. A method for adjusting spatial congruency in a video conference is disclosed. The method includes detecting spatial congruency between a visual scene captured by a video endpoint device and an auditory scene captured by an audio endpoint device that is positioned in relation to the video endpoint device, the spatial congruency being a degree of alignment between the auditory scene and the visual scene, comparing the detected spatial congruency with a predefined threshold and in response to the detected spatial congruency being below the threshold, adjusting the spatial congruency. Corresponding system and computer program products are also disclosed.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Abstract: A method of post-processing banded gains for applying to an audio signal, an apparatus to post-processed banded gains, and a tangible computer-readable storage medium comprising instructions that when executed carry out the method. The banded gains are determined by input processing one or more input audio signals. The method includes post-processing the banded gains to generate post-processed gains, generating a particular post-processed gain for a particular frequency band including percentile filtering using gain values from one or more previous frames of the one or more input audio signals and from gain values for frequency bands adjacent to the particular frequency band.

Abstract: Example embodiments disclosed herein relate to user experience oriented audio signal processing. There is provided a method for user experience oriented audio signal processing. The method includes obtaining a first audio signal from an audio sensor of an electronic device; computing, based on the first audio signal, a compensation factor for an acoustic path from the electronic device to a listener and applying the compensation factor to a second audio signal outputted from the electronic device. Corresponding system and computer program products are disclosed.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: Embodiments of the present invention relate to adaptive audio content generation. Specifically, a method for generating adaptive audio content is provided. The method comprises extracting at least one audio object from channel-based source audio content, and generating the adaptive audio content at least partially based on the at least one audio object. Corresponding system and computer program product are also disclosed.

Abstract: A method for controlling acoustic echo cancellation and an audio processing apparatus are described. In one embodiment, the audio processing apparatus includes an acoustic echo canceller for suppressing acoustic echo in a microphone signal, a jitter buffer for reducing delay jitter of a received signal, and a joint controller for controlling the acoustic echo canceller by referring to at least one future frame in the jitter buffer.

Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that results from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets (412, 413) is described. A lost packet (412) is a packet which is deemed to be lost by a transform-based audio decoder. Each of the one or more lost packets (412, 413) comprises a set of transform coefficients (313). A set of transform coefficients (313) is used by the transform-based audio decoder to generate a corresponding frame (412, 413) of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet (412) of the one or more lost packets (412, 413) a number of preceding lost packets from the one or more lost packets (313); wherein the determined number is referred to as a loss position.