摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
A sound processor integrated on a single chip with multiple digital sound sample stream inputs. Each input is independently connected to separate ports of a multi-port memory. The architecture allows multiple, asynchronous digital sound sample streams to be concurrently loaded into the memory without requiring synchronization to any particular input stream.
摘要:
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.
摘要:
An audio data format in which an instrument is described using a combination of sound samples and articulation instructions which determine modifications made to the sound sample is provided. The instruments form a first, initial layer, with a second layer having presets which can user defined to provide additional articulation instructions which can modify the articulation instructions at the instrument level. The articulation instructions are specified using various parameters. The present invention provides a format in which all of the parameters are specified in units which relate to a physical phenomena, and thus are not tied to any particular machine for creating or playing the audio samples. The articulation parameters include generators and modulators, which provide a connection between a real-time signal and a generator. The parameter units are specified in perceptually additive units, to make the data portable and easily edited. New units are defined to give perceptual additive parameters throughout.
摘要:
An improved dynamic digital IIR (Infinite Impulse Response) audio filter and method provides a remedy for the major problems in state of the art digital IIR filtering and dynamic digital filtering for audio. In particular, the required multiplier coefficient size is reduced, such that 16 bit coefficients are adequate for good frequency resolution. In addition, the data size of the multiplier is also reduced, to be similar to the input and output data resolution without inferior noise performance. Furthermore, the structure is easily modified to allow the fixing of the DC (Direct Current) gain of the IIR filter to unity regardless of coefficient choice, eliminating any problems with filter dynamic range at low frequency passbands. Additionally, a dynamic filter structure allows logarithmic filter sweeps and compresses the coefficients further down to as few as 8 bits with adequate resolution for audio throughout the band. The structure allows the alteration of sweep of the filter without excessive computation or the requirement to read the current sweep value.
摘要:
An output stage for a multitimbral electronic musical instrument providing automatic detection of the use of submix outputs is provided. The present invention allows the use of effects processors on selected timbres without the need for user intervention when the effects are connected or removed. The present invention also allows for use of such processors without an external mixboard.
摘要:
A lowpass filter circuit having an electronically controllable cutoff frequency and utilizing a feedback current mirror circuit as a variable impedance element. The filter circuit employs low noise, low distortion output circuitry and includes means for electronically controlling resonance.