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:
A method for processing audio data, the method comprising: receiving audio data corresponding to a plurality of instances of audio, including at least one of: (a) audio data from multiple endpoints, recorded separately or (b) audio data from a single endpoint corresponding to multiple talkers and including spatial information for each of the multiple talkers; rendering the audio data in a virtual acoustic space such that each of the instances of audio has a respective different virtual position in the virtual acoustic space; and scheduling the instances of audio to be played back with a playback overlap between at least two of the instances of audio, wherein the scheduling is performed, at least in part, according to a set of perceptually-motivated rules.
Abstract:
Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.
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:
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:
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:
Embodiments of the present invention relate to speaker identification using spatial information. A method of speaker identification for audio content being of a format based on multiple channels is disclosed. The method comprises extracting, from a first audio clip in the format, a plurality of spatial acoustic features across the multiple channels and location information, the first audio clip containing voices from a speaker, and constructing a first model for the speaker based on the spatial acoustic features and the location information, the first model indicating a characteristic of the voices from the speaker. The method further comprises identifying whether the audio content contains voices from the speaker based on the first model. Corresponding system and computer program product are also disclosed.
Abstract:
A method of encoding audio information for forward error correction reconstruction of a transmitted audio stream over a lossy packet switched network, the method including the steps of: (a) dividing the audio stream into audio frames; (b) determining a series of corresponding audio frequency bands for the audio frames; (c) determining a series of power envelopes for the frequency bands; (d) encoding the envelopes as a low bit rate version of the audio frame in a redundant transmission frame.
Abstract:
Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.
Abstract:
An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method. By assigning different hearing properties to desired sound and noise, the intelligibility of the audio signal can be improved.