Abstract: Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Abstract: Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Abstract: Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Abstract: At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Abstract: At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Abstract: At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Abstract: Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Abstract: Provided are methods and systems for reducing a transition rate in transmitting data between analog and digital chips in Sigma-Delta Modulator (SDM) based Digital to Analog Converters (DACs) and Analog to Digital Converters (ADCs) intended to be used in audio signal processing. An example method may comprise receiving, by a digital chip, SDM binary data, mapping the SDM binary data to transition binary codes, and transmitting the transition binary codes to an analog chip. The mapping can be carried out according to a principle that the more commonly used SDM binary data codes correspond to transition binary data codes that require that fewer transitions occur in the signals between the chips. The methods and systems described provide for lowering the power needed for carrying out the data transmission between digital and analog chips.

Abstract: Systems and methods for sample rate tracking are provided. An example method includes computing an actual latency associated with an output sample from an output sample stream. The actual latency is calculated using a phase and a phase increment (conversion rate ratio). A measured latency is determined using an internal clock using a presentation time of the output sample, or an input sample from an input sample stream, or both. The measured latency is compared to the actual latency to generate a latency error. A successive phase increment can be determined based on the latency error by using a low-pass or adaptive filter to adjust the latency error.

Abstract: Systems and methods for voice-controlled communication connections are provided. An example system includes a mobile device being operated consecutively in listen, wakeup, authentication, and connect modes. Each of subsequent modes consumes more power than a preceding mode. The listen mode consumes less than 5 mW. In the listen mode, the mobile device listens for an acoustic signal, determines whether the acoustic signal includes voice, and upon the determination, selectively enters the wakeup mode. In the wakeup mode, the mobile device determines whether the acoustic signal includes a spoken word and, upon the determination, enters the authentication mode. In authentication mode, the mobile device identifies a user using the spoken command and, upon the identification, enters the connect mode. In the connect mode, the mobile device receives an acoustic signal, determines whether the acoustic signal includes a spoken command and performs one or more operations associated with the spoken command.

Abstract: Digital methods and systems for signal processing and filtering are provided. The methods and corresponding systems provide asynchronous conversion of sampling rate frequencies and utilize advanced multistage phasor filters for converting an input signal having a first sampling rate into an output signal sampled in an arbitrary sequence of sampling times. The conversion process provides a sequence of sets of complex numbers representing a filtered version of the input signal. More specifically, the conversion process includes the calculation of values of the output signal by multiplying (e.g., scaling) the sets of complex numbers by a corresponding set of complex phasors, the complex phasors corresponding to the timing of the arbitrary time sequence to obtain a corresponding set of real results with the value of the output signal being the sum of the real results.

Abstract: A method and a computer program product for sample rate conversion that features distributive or hybrid filtering to reduce unwanted artifacts, such as aliasing and the computational requirements to avoid the aforementioned artifacts. The method includes receiving, at a first sample rate, a plurality of data points, associated with a first signal, operating on the plurality of data points to associate the signal with a predetermined set of parameters, with the set of parameters including a first transition band having an image associated therewith; and varying the sample rate associated with the first signal by interpolation with an interpolator having associated therewith a second transition band, with the width associated with the second transition band being a function of a spectral separation between the first transition band and its image, wherein a second signal is produced having a sequence of data samples approximating the first signal.

Abstract: A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.

Abstract: A cache memory is updated with audio samples in a manner which minimizes system bus bandwidth and cache size requirements. The end of a loop is used to truncate a normal cache request to exactly what is needed. A channel with a loopEnd in a request will be given higher priority in a two-stage priority scheme. The requested data is conformed by trimming to the minimum data block size of the bus, such a doubleword for a PCI bus. The audio data written into the cache can be shifted on a byte-wise basis, and unneeded bytes can be blocked and not written. Request data for which a bus request has been issued can be preempted by request data attaining a higher priority before a bus grant is received.

Abstract: A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.

Abstract: A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.

Abstract: A cache memory is updated with audio samples in a manner which minimizes system bus bandwidth and cache size requirements. The end of a loop is used to truncate a normal cache request to exactly what is needed. A channel with a loopEnd in a request will be given higher priority in a two-stage priority scheme. The requested data is conformed by trimming to the minimum data block size of the bus, such a doubleword for a PCI bus. The audio data written into the cache can be shifted on a byte-wise basis, and unneeded bytes can be blocked and not written. Request data for which a bus request has been issued can be preempted by request data attaining a higher priority before a bus grant is received.

Abstract: A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.

Abstract: An asynchronous sample rate tracker with rapid acquisition and good steady state performance is provided. In one embodiment, dual tracking loops are used to control reading from a FIFO sample buffer and generation of a ratio of source and local sampling rates. One tracking loop is used for rapid acquisition when the buffer is either empty or full with another tracking loop being used for steady-state operation. The steady-state tracking loop incorporates a low-pass filter to remove the effects of momentary variations in source sampling rates.

Abstract: A circuit and method for generating waveforms when synthesizing musical sounds. In one embodiment, the invention provides a multiplexer/shifter which modifies the phase angle input according to the particular waveform desired. Boolean logic gates further modify the multiplexer/shifter output based on the two most significant bits of the phase angle input and according to the particular waveform desired. Finally, a multiplier multiplies the multiplexer/shifter output with the output of the Boolean logic gates to produce the desired waveform. The invention may employ banks of exclusive OR gates and AND gates as the Boolean logic. Another embodiment of the invention provides a waveshaping method where a desired waveform is generated from a phase angle input. The phase angle input is multiplexed/shifted based on the particular waveform desired.