Abstract: In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Abstract: In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Abstract: Noise, either in the form of comfort noise or film grain, is added to a three dimensional image in accordance with image depth information to reduce human sensitivity to coding artifacts, thereby improving subjective image quality.

Abstract: A continuous comfort noise is provided that is overlaid for the entire duration of a conference call scenario. The comfort noise may be adapted to match the levels of the actual background noise detected on one or more of the conference call participant's devices on the transmitting end(s) of a conference call as well as the participants' speech levels. The comfort noise may also be adapted to the type of listening device employed on the receiving end of a conference call. The comfort noise level may be customized to an appropriate and comfortable level for the type of listening device being used, and the system may continuously mix the comfort noise with incoming audio signals for the entire duration of a conference call, lowering the comfort noise level gradually during speaking periods for additional user experience improvement.

Abstract: A method, medium, and system scalably encoding/decoding audio/speech. The method includes splitting an input signal into a low frequency band signal that is lower than a predetermined frequency and a high frequency band signal that is higher than the predetermined frequency, scalably encoding the split low frequency band signal into a core layer and one or more extension layers and then decoding the encoded core layer and the encoded extension layers, generating an error signal by using the split low frequency band signal and a decoded signal of the encoded core layer and the encoded extension layers, and encoding the error signal and the high frequency band signal into a signal-to-noise ratio (SNR) enhancement layer and a bandwidth extension layer.

Abstract: A method and apparatus for performing speech quality assessment in a speech communication system (such as, for example, a VoIP communication system) which detects and measures the presence of impulsive noise is provided. Specifically, in one illustrative embodiment, an autoregressive (AR) model of speech (and, in particular, of the excitation of the vocal tract) is advantageously employed to estimate a short-term variance of the speech excitation, and the standard deviation of the speech excitation (i.e., the square root of the variance) is then advantageously compared to a predetermined threshold to identify whether impulsive noise is present. Then, based on a statistic analysis of any such identified impulsive noise, a speech quality assessment is generated.

Abstract: A voice input device includes a first microphone (710-1) that includes a first diaphragm, a second microphone (710-2) that includes a second diaphragm, and a differential signal generation section (720) that generates a differential signal that indicates a difference between a first voltage signal and a second voltage signal, the first diaphragm and the second diaphragm being disposed so that a noise intensity ratio is smaller than an input voice intensity ratio (input voice component intensity ratio), and the differential signal generation section (720) including a gain section (760) that amplifies the first voltage signal by a predetermined gain, and a differential signal output section (740) that generates and outputs a differential signal that indicates a difference between the first voltage signal amplified by the gain section and the second voltage signal.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: A transform coder leading to reduction of degradation of perceptual sound quality even if an adequate number of bits is not assigned. Candidates of a correction scale factor stored in a correction scale factor codebook (123) are outputted one by one, and an error signal is generated by subjecting the candidate and scale factors outputted from scale factor computing sections (121, 122) to a predetermined operation. A judging section (126) determines a weight vector given to a weighted error computing section (127) depending on the sign of the error signal. The weighted error computing section (127) computes the square of the error signal, multiplies the square of the error signal by the weight vector given from the judging section (126), and computes a weighted squared error E. A search section (128) determines the candidates of the correction scale factor which minimizes the weighted squared error E by a closed loop processing.

Abstract: A device comprising an audio information processor to receive at least one audio stream encoded according to a first protocol by a remote network processing device, the audio stream having associated comfort noise information to indicate a level of background noise available for presentation during silence periods associated with the audio stream, the audio information processor to decode the received audio stream according to the first protocol and to encode the decoded audio stream according to a second protocol, and a background noise translator to convert the comfort noise information received with the audio stream into a format compatible with the second protocol.

Abstract: A system, method, and apparatus for separating speech signal from a noisy acoustic environment. The separation process may include directional filtering, blind source separation, and dual input spectral subtraction noise suppressor. The input channels may include two omnidirectional microphones whose output is processed using phase delay filtering to form speech and noise beamforms. Further, the beamforms may be frequency corrected. The omnidirectional microphones generate one channel that is substantially only noise, and another channel that is a combination of noise and speech. A blind source separation algorithm augments the directional separation through statistical techniques. The noise signal and speech signal are then used to set process characteristics at a dual input noise spectral subtraction suppressor (DINS) to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify speech signal.

Abstract: A method and system for improving speech quality may include estimating at least one component of a distorted portion of a speech signal from at least one component of an undistorted portion of the speech signal and reinforcing the component of the distorted portion based on the estimating. The components may include the pitch, spectral envelope and spectral energy of the speech signal. The undistorted portion of the speech signal may be delayed and the components of the distorted portion may be interpolated from the components of a delayed undistorted portion and a current undistorted portion of the speech signal. The components of the distorted portion of the speech signal may be extrapolated from a current undistorted portion of the speech signal. Components of the distorted portion of the speech signal may be estimated from frequency bands other than the frequency band affected by the distortion.

Abstract: In one of many possible embodiments, a method includes providing an audio output signal to an output device for broadcast to a user, receiving audio input, the audio input including user voice input provided by the user and audio content broadcast by the output device in response to receiving the audio output signal, applying at least one predetermined calibration setting, and filtering the audio input based on the audio output signal and the predetermined calibration setting. In some examples, the calibration setting may be determined in advance by providing a calibration audio output signal to the output device for broadcast, receiving calibration audio input, the calibration audio input including calibration audio content broadcast by the output device in response to receiving the calibration audio output signal, and determining the calibration setting based on at least one difference between the calibration audio output signal and the calibration audio input.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal.