Abstract:
Compressibility-based reallocation of initial bit allocations for frames of an audio signal is described. Applications to redundancy-based retransmission of critical frames (e.g., for fixed-bit-rate modes of speech codec operation) are also described.
Abstract:
A method for detecting overflow on an electronic device is described. The method includes determining a linear predictive coding synthesis filter gain. The method further includes determining whether overflow is detected based on the linear predictive coding synthesis filter gain and a fixed codebook gain. The method further includes determining a scaling factor if overflow is detected.
Abstract:
A method and an apparatus for estimating speech signal in split-domain is disclosed. The method includes performing LP analysis on a noisy speech signal to generate a first plurality of LPC and a first residual signal. The method also includes estimating speech LPC spectrum to generate cleaned LPC. The method further includes estimating speech residual spectrum to generate cleaned residual signal. The method also includes synthesizing output signals based on the cleaned LPC and the cleaned residual signal.
Abstract:
A method of capturing audio includes initiating capture, at a laser microphone, of first audio of an area of interest. The first audio is captured while the laser microphone is focused on a first target surface associated with the area of interest. The method also includes generating adjustment parameters based on a feedback signal to adjust targeting characteristics of the laser microphone. The method further includes adjusting the targeting characteristics of the laser microphone based on the adjustment parameters to focus the laser microphone on a second target surface associated with the area of interest. The method also includes initiating capture, at the laser microphone, of second audio of the area of interest in response to adjusting the targeting characteristics. The second audio has an audio quality that is greater than the first audio.
Abstract:
A method includes extracting a voicing classification parameter of an audio signal and determining a filter coefficient of a low pass filter based on the voicing classification parameter. The method also includes filtering a low-band portion of the audio signal to generate a low-band audio signal and controlling an amplitude of a temporal envelope of the low-band audio signal based on the filter coefficient. The method also includes modulating a white noise signal based on the amplitude of the temporal envelope to generate a modulated white noise signal and scaling the modulated white noise signal based on a noise gain to generate a scaled modulated white noise signal. The method also includes mixing a scaled version of the low-band audio signal with the scaled modulated white noise signal to generate a high-band excitation signal that is used to generate a decoded version of the audio signal.
Abstract:
A device including gain shape circuitry configured to determine a number of sub-frames of multiple sub-frames that are saturated, the multiple sub-frames included in a frame of a high band audio signal. The device also includes gain frame circuitry configured to determine, based on the number of sub-frames that are saturated, a gain frame parameter corresponding to the frame.
Abstract:
An apparatus includes a first calculator configured to determine a long-term noise estimate of the audio signal. The apparatus also includes a second calculator configured to determine a formant-sharpening factor based on the determined long-term noise estimate. The apparatus includes a filter configured to filter a codebook vector to generate a filtered codebook vector. The filter is based on the determined formant-sharpening factor, and the codebook vector is based on information from the audio signal. The apparatus further includes an audio coder configured to generate a formant-sharpened low-band excitation signal based on the filtered codebook vector.
Abstract:
A device includes a receiver, a buffer, a transmitter, and an analyzer. The receiver is configured to receive a plurality of packets that corresponds to at least a subset of a sequence of packets. Error correction data of a first packet of the plurality of packets includes a partial copy of a second packet of the plurality of packets. The analyzer is configured to determine whether a particular packet of the sequence is missing from the buffer, and to determine whether a partial copy of the particular packet is stored in the buffer. The analyzer is also configured to send, via the transmitter, a retransmit message to a second device based at least in part on determining that the buffer does not store the particular packet and that the buffer does not store the partial copy of the particular packet.
Abstract:
A method for adjusting a delay of a buffer at a receiving terminal includes determining, at a processor, a partial frame recovery rate of lost frames at the receiving terminal. The method also includes adjusting the delay of the buffer based at least in part on the partial frame recovery rate.
Abstract:
A method includes determining, at a decoder of a first device, an offset value corresponding to an offset between a first particular packet and a second particular packet. The first device includes a de-jitter buffer. The method also includes transmitting the offset value to an encoder of a second device to enable the second device to send packets to the first device based on the offset value.