Patent classifications
G10L21/0316
SIGNAL PROCESSING METHOD AND DEVICE
A signal processing method and device are provided. At least two channel sound signals are acquired, and a frequency-domain audio signal corresponding to each channel sound signal is acquired; beam forming output signals of a beam group corresponding to an audio signal of each frequency point are acquired; an output direction of the beam group is acquired; and time-domain sound signals output after beam forming in the output direction are acquired.
Audio visual correspondence based signal augmentation
A system includes a headset to capture sound and a visual signal of a local area including one or more sound sources. The system determines a strength of the audio signal and a portion of the visual signal associated with the audio signal, compares the strengths, selects the weaker signal, and augments the weaker signal. The headset accordingly presents augmented audio-visual content to a user, thereby enhancing the user's perception of the weak signal.
Audio visual correspondence based signal augmentation
A system includes a headset to capture sound and a visual signal of a local area including one or more sound sources. The system determines a strength of the audio signal and a portion of the visual signal associated with the audio signal, compares the strengths, selects the weaker signal, and augments the weaker signal. The headset accordingly presents augmented audio-visual content to a user, thereby enhancing the user's perception of the weak signal.
Method, device and software for controlling transport of audio data
A method for processing music audio data, including providing input audio data representing a first piece of music comprising a mixture of musical timbres. The method also includes decomposing the input audio data to generate at least first-timbre decomposed data representing a first timbre selected from the musical timbres of the first piece of music, and second-timbre decomposed data representing a second timbre selected from the musical timbres of the first piece of music. The method also includes applying a transport control to obtain transport controlled first-timbre decomposed data. The method also includes recombining audio data obtained from the transport controlled first-timbre decomposed data with audio data obtained from the second-timbre decomposed data to obtain recombined audio data.
DECODING METHOD AND DECODER FOR DIALOG ENHANCEMENT
There is provided a method for enhancing dialog in a decoder of an audio system. The method comprises receiving a plurality of downmix signals being a downmix of a larger plurality of channels; receiving parameters for dialog enhancement being defined with respect to a subset of the plurality of channels that is downmixed into a subset of the plurality of downmix signals; upmixing the subset of downmix signals parametrically in order to reconstruct the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined; applying dialog enhancement to the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined using the parameters for dialog enhancement to provide at least one dialog enhanced signal; and subjecting the at least one dialog enhanced signal to mixing to provide dialog enhanced versions of the subset of downmix signals.
VEHICLE AND CONTROL METHOD THEREOF
A vehicle and control method of the vehicle are provided. The vehicle includes a camera provided on the vehicle and configured to capture an image of an object outside the vehicle, a controller configured to determine a photographing position required for facial recognition from the captured image, a guide configured to guide the photographing position, and a display configured to display a result of the facial recognition.
Rate convertor
Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.
Rate convertor
Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.
Signal processing apparatus, signal processing method and non-transitory computer-readable recording medium
A Finite Impulse Response (FIR) filter is configured to minimize delay and maximize passband power by adjusting the filter coefficients applied to the sampled values. The FIR filter obtains an input signal and samples the input signal to generate a set of sampled input values. The FIR filter generates a set of filter coefficients, with each filter coefficient based on a corresponding sampled input value in the set of sample input values. The FIR filter selects a subset of sampled input values that have been most recently sampled from the input signal, and selects a subset of filter coefficients corresponding to sampled input values that are not the most recently sampled. The subset of sampled input values is combined with the subset of filter coefficients to generate an output value for the FIR filter.
VOICE WAKE-UP METHOD AND DEVICE
Disclosed is a terminal, comprising a microphone, a power supply switching control circuit, a data switching control circuit, a voice wake-up circuit and a voice wake-up power supply; the microphone is configured to acquire a voice signal and input the voice signal to the data switching control circuit; the power supply switching control circuit is configured to supply power for the microphone via the voice wake-up power supply when receiving a voice wake-up instruction; the data switching control circuit is configured to input the voice signal to the voice wake-up circuit when receiving the voice wake-up instruction; and the voice wake-up circuit is configured to preprocess and match the input voice signal, and execute a corresponding operation according to a matching result.
Also disclosed is a voice wake-up method.