Abstract: Provided are methods and systems for providing situation-dependent transient noise suppression for audio signals. Different strategies (e.g., levels of aggressiveness) of transient suppression and signal restoration are applied to audio signals associated with participants in a video/audio conference depending on whether or not each participant is speaking (e.g., whether a voiced segment or an unvoiced/non-speech segment of audio is present). If no participants are speaking or there is an unvoiced/non-speech sound present, a more aggressive strategy for transient suppression and signal restoration is utilized. On the other hand, where voiced audio is detected (e.g., a participant is speaking), the methods and systems apply a softer, less aggressive suppression and restoration process.

Abstract: A system includes a speaker, an acoustic echo canceller, a post-processor configured to create a post-processed render signal associated with an audio input, and a reference path operatively connected to the speaker, the post-processor, and the acoustic echo canceller. The reference path provides the acoustic echo canceller with access to the post-processed render signal.

Abstract: An apparatus provides both clipping protection and signal level conservation while the system operates in the original width type. The apparatus includes a first shifting unit right shifting a first digital input signal to provide a first shifted signal; a second shifting unit right shifting a second digital input signal to provide a second shifted signal; a combiner combining the first shifted signal and the second shifted signal to provide a combined signal; a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal; and a third shifting unit left shifting the soft limited signal to provide an output signal.

Abstract: Methods and systems are provided for detecting chop in an audio signal. A time-frequency representation, such as a spectrogram, is created for an audio signal and used to calculate a gradient of mean power per frame of the audio signal. Positive and negative gradients are defined for the signal based on the gradient of mean power, and a maximum overlap offset between the positive and negative gradients is determined by calculating a value that maximizes the cross-correlation of the positive and negative gradients. The negative gradient values may be combined (e.g., summed) with the overlap offset, and the combined values then compared with a threshold to estimate the amount of chop present in the audio signal. The chop detection model provided is low-complexity and is applicable to narrowband, wideband, and superwideband speech.

Abstract: Methods and systems are provided for using a model of human speech quality perception to provide an objective measure for predicting subjective quality assessments. A Virtual Speech Quality Objective Listener (ViSQOL) model is a signal-based full-reference metric that uses a spectro-temporal measure of similarity between a reference signal and test speech signal. Specifically, the model provides for the ability to detect and predict the level of clock drift, and determine whether such clock drift will impact a listener's quality of experience.

Abstract: An apparatus provides both clipping protection and signal level conservation while the system operates in the original width type. The apparatus includes a first shifting unit right shifting a first digital input signal to provide a first shifted signal; a second shifting unit right shifting a second digital input signal to provide a second shifted signal; a combiner combining the first shifted signal and the second shifted signal to provide a combined signal; a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal; and a third shifting unit left shifting the soft limited signal to provide an output signal.

Abstract: Methods and systems for using pitch predictors in speech/audio coders are provided. Techniques for optimal pre- and post-filtering are presented, and a general result that post-filtering is more effective than pre-filtering is derived. A practical paired-zero filter design for the low-rate regime is proposed, and this design is extended to handle frequency-dependent periodicity levels. Further, the methods described provide a general performance measure for a post-filter that only uses information available at the decoder, thereby allowing for the optimization or selection of a post-filter without increasing the rate.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: A mixing system provides both clipping protection and signal level conservation while the system operates in the original width type. The mixing system includes a first input multiplier multiplying a first digital input signal by a first gain value to provide a first scaled signal, a second input multiplier multiplying a second digital input signal by the first gain value to provide a second scaled signal, a combiner combining the first scaled signal and the second scaled signal to provide a combined signal, a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal, and an output multiplier multiplying the soft limited signal by a second gain value to provide a mixed output signal, wherein the first gain value is a value that is equal to an inverse value of the second gain value.

Abstract: Provided are methods and systems for detecting the presence of a transient noise event in an audio stream using primarily or exclusively the incoming audio data. Such an approach offers improved temporal resolution and is computationally efficient. The methods and systems presented utilize some time-frequency representation of an audio signal as the basis in a predictive model in an attempt to find outlying transient noise events and interpret the true detection state as a Hidden Markov Model (HMM) to model temporal and frequency cohesion common amongst transient noise events.

Abstract: A method and an apparatus for indicating presence of a transient noise in a call are provided. The method comprises the steps of determining activity at an endpoint of the call by monitoring presence of a signal input from the endpoint into the call and monitoring presence of a potential source of transient noise at the endpoint. Further, based on the activity determination and the monitoring of the presence of a potential source of transient noise, a signal representative of the presence of a transient noise in the call is sent. The present invention is advantageous in that it enables improvement of the quality of the call.

Abstract: Methods and systems for detecting the presence and frequency of clipping in an audio signal are provided. A clipping detection algorithm detects the presence of hard and soft clipping using histograms with intervals of samples, rather than attempting to identify the clipping value. Therefore, it is not essential to the algorithm that there be a large number of bins. Furthermore, the bins may be non-uniformly distributed since the number of samples belonging to lower amplitudes is of little importance. The detection algorithm is also configured to determine the severity and/or perceptual effect of any clipping found to be present in the signal by calculating the ratio of clipped samples to non-clipped samples. Temporal information on the occurrence of clipping in the signal is also used to evaluate perceptual effect.

Abstract: Provided are methods and systems for rearranging a multichannel audio signal into sub-signals and allocating bit rates among them, such that compressing the sub-signals with a set of audio codecs at the allocated bit rates yields an optimal fidelity with respect to the original multichannel audio signal. Rearranging the multichannel audio signal into sub-signals and assigning each sub-signal a bit rate may be optimized according to a criterion. Existing audio codecs may be used to quantize the sub-signals at the assigned bit rates and the compressed sub-signals may be combined into the original format according to the manner in which the original multichannel audio signal is rearranged.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: A mixing system provides both clipping protection and signal level conservation while the system operates in the original width type. The mixing system includes a first input multiplier multiplying a first digital input signal by a first gain value to provide a first scaled signal, a second input multiplier multiplying a second digital input signal by the first gain value to provide a second scaled signal, a combiner combining the first scaled signal and the second scaled signal to provide a combined signal, a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal, and an output multiplier multiplying the soft limited signal by a second gain value to provide a mixed output signal, wherein the first gain value is a value that is equal to an inverse value of the second gain value.

Abstract: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.

Abstract: Processing multi-channel audio streams using one or more arrangements of single-channel components. Components that only process the near-end, or capture stream, such as noise suppression (NS) components, are limited in how they can be suitably arranged for processing multi-channel streams. However, components that process the near-end stream using one or more inputs from the far-end, or render stream, such as acoustic echo cancellation (AEC) and automatic gain control (AGC) components, are arranged in one or more of the ways suitable for use with multiple channels.