Abstract: Automatically generating a hardware image based on programming model types includes determining by a design tool, types of programming models used in specifications of blocks of a circuit design, in response to a user control input to generate a hardware image to configure a programmable integrated circuit (IC). The design tool can generate a model-type compiler script for each of the types of programming models. Each compiler script initiates compilation of blocks having specifications based on one of the types of programming model into an accelerator representation. The design tool can generate a build script configured to execute the compiler scripts and link the accelerator representations into linked accelerator representations. Execution of the build script builds a hardware image from the linked accelerator representations for configuring the programmable IC to implement a circuit according to the circuit design.

Abstract: Model-based implementation of a design for a heterogeneous integrated circuit can include converting a model, created as a data structure using a modeling system, into a data flow graph, wherein the model represents a design for implementation in an integrated circuit having a plurality of systems, the systems being heterogeneous, classifying nodes of the data flow graph for implementation in different ones of the plurality of systems of the integrated circuit, and partitioning the data flow graph into a plurality of sub-graphs based on the classifying, wherein each sub-graph corresponds to a different one of the plurality of systems. From each sub-graph, a portion of high-level language (HLL) program code can be generated. Each portion of HLL program code may be specific to the system corresponding to the sub-graph from which the portion of HLL program code was generated.

Abstract: Approaches for simulating and processing a circuit design involve recognizing by a design processing tool a replaceable subsystem in a circuit design having multiple blocks. The replaceable subsystem includes a subset of the blocks. The design tool converts the subset of blocks into an executable program and schedules activation of blocks of the circuit design other than the subset of blocks during simulation of the circuit design. The scheduled blocks are activated during simulation according to the scheduling, and activation of the subset of the plurality of blocks is bypassed during simulation with a call to the executable program.

Abstract: The disclosed methods, mechanisms, and system provide the ability to associate one or more disjointed models such that a second model can be accessed from a first model thereby allowing easier navigation between models. A graphical link to a second model is provided as part of a first model. Activating the graphical link opens the second model for viewing or editing. Links between additional models may also be provided. Thus, by using the provided graphical links, a user may quickly and easily navigate between all the associated models.

Abstract: The disclosed methods, mechanisms, and system provide the ability to associate one or more disjointed models such that a second model can be accessed from a first model thereby allowing easier navigation between models. A graphical link to a second model is provided as part of a first model. Activating the graphical link opens the second model for viewing or editing. Links between additional models may also be provided. Thus, by using the provided graphical links, a user may quickly and easily navigate between all the associated models.

Abstract: The invention provides a method and system for dynamically estimating background noise. The system includes a portable communication device, a vocoder, and a voice activated detector. Based on information received by the portable communication device, the vocoder determines parameters related to incoming information including a voicing mode indicative of the periodicity of incoming information. The voice activated detector then compares the voicing mode to a threshold to determine whether a background noise estimate should be updated.

Abstract: A mobile communication device (100) includes a vocoder (104) for vocoding speech (500) received at the mobile communication device. The parameters output by the vocoder are used to generate a voicing quality metric (208). The voicing quality metric is used to provide feedback to the user of the mobile communication device by various feedback modalities including visual (114), audible (108), and tactile modalities (118) to indicate when the user should speak louder to overcome ambient noise. The voicing quality metric is also used by other communications equipment (304, 312) to decide if communication activity is needed.

Abstract: A sigma delta digital-to-analog (D/A) converter system (10) includes a summing device (35) at an input of a D/A converter (30), and a low frequency low amplitude wave signal (31) injected at an input of the summing device that remains unfiltered and is used to suppress spurious tone artifacts. The D/A converter system can further include an amplitude control and a frequency control for selectively adjusting the frequency and the amplitude of the low frequency low amplitude wave signal being injected. Note, the low frequency low amplitude repeating wave signal generator can take the form of a digital signal processor (DSP) (37) having the appropriate software to generate such signals.

Abstract: A communication device (200) for dual mode muting operation includes a user interface (208) and a controller (202). The controller (202) is programmed to: in response to a first user input to the user interface (208), enable a first muting mode; and in response to a second user input to the user interface (208), disable the first muting mode and enable a second muting mode. The first muting mode may be one of a full muting and a concealed muting, while the second muting mode may be the other one of a full muting and a concealed muting.

Abstract: A sigma delta digital-to-analog (D/A) converter system (10) includes a summing device (35) at an input of a D/A converter (30), and a low frequency low amplitude wave signal (31) injected at an input of the summing device that remains unfiltered and is used to suppress spurious tone artifacts. The D/A converter system can further include an amplitude control and a frequency control for selectively adjusting the frequency and the amplitude of the low frequency low amplitude wave signal being injected. Note, the low frequency low amplitude repeating wave signal generator can take the form of a digital signal processor (DSP) (37) having the appropriate software to generate such signals.

Abstract: A telephone (310) and a method for providing outbound audio when the telephone is operating in a speakerphone mode. A first data unit (350) including a first unit type identifier (360) can be received by the telephone. The first unit type identifier can be an indicator of a type of audio data contained in the first data unit. For instance, the first unit type identifier can indicate whether the audio data is music or non-music audio data. If the first unit type identifier has a first value, for example a value indicating that the audio data is music data, unmuted outbound audio reproduced from the first data unit can be provided and voice activity detection can be disabled. Additionally, inbound audio can be muted.

Abstract: The invention concerns a method (300) and system (100) for improving voice quality of a vocoder (138, 158). The method includes the steps of monitoring (312) a pitch of a voice signal (400) at a transmitting unit (110); when the pitch of the voice signal reaches a predetermined threshold (840), shifting (326) the pitch of the voice signal to at least a portion of a predetermined range (810); transmitting (338) the pitch-shifted voice signal to a receiving unit (112); and at the receiving unit, reshifting (342) the pitch-shifted voice signal to a level that compensates the step of shifting the pitch of the voice signal at the transmitting unit.

Abstract: The invention concerns a method (300) and system (100) for controlling audio output. The method includes the steps of inputting (312) an audio signal and a voltage level signal, measuring (314) the audio signal and the voltage level signal, mapping (316) the audio signal against at least one table (134) of predetermined corresponding gain targets (138) and selecting (318) at least one gain target for the audio signal. The mapping step and the selecting step are based at least in part on the measurement of the voltage level signal and the measurement of the audio signal. The method also includes the step of applying (320) the gain target to the audio signal.

Abstract: A hands-free digital push-to-talk device (102) includes a digital background noise suppressor (302), a digital voice activity detector (304), an audio buffer (306), as well as a decision handler (308), embedded inside the device's (102) digital signal processor (222). Audio is buffered until the decision handler (308) determines that speech is present on an audio stream fed to the voice activity detector (304). The decision handler (308) makes the decision by assigning weighted values to each voice activity detector (304) determination, the weighted value varying depending on the state of the device (102) and temporal distance from the present time.

Abstract: Systems, an apparatus, and methods are provided for mitigating noise associated with an audio signal. A system (100) for mitigating noise associated with an audio signal includes an estimator module (108). The estimator module determines an estimated level of noise associated with the audio signal. The system also includes an expander module (110). The expander module causes an attenuation of the audio signal if a level of the audio signal is below a signal threshold. The expander module is adaptively tunable so that the attenuation caused (606) by the expander module is based upon the level of noise estimated (602) by the estimator module.

Abstract: A telephone (310) and a method for providing outbound audio when the telephone is operating in a speakerphone mode. A first data unit (350) including a first unit type identifier (360) can be received by the telephone. The first unit type identifier can be an indicator of a type of audio data contained in the first data unit. For instance, the first unit type identifier can indicate whether the audio data is music or non-music audio data. If the first unit type identifier has a first value, for example a value indicating that the audio data is music data, unmuted outbound audio reproduced from the first data unit can be provided and voice activity detection can be disabled. Additionally, inbound audio can be muted.

Abstract: The invention concerns a method (300) and system (100) for improving voice quality of a vocoder (138, 158). The method includes the steps of monitoring (312) a pitch of a voice signal (400) at a transmitting unit (110); when the pitch of the voice signal reaches a predetermined threshold (840), shifting (326) the pitch of the voice signal to at least a portion of a predetermined range (810); transmitting (338) the pitch-shifted voice signal to a receiving unit (112); and at the receiving unit, reshifting (342) the pitch-shifted voice signal to a level that compensates the step of shifting the pitch of the voice signal at the transmitting unit.

Abstract: The invention concerns a method (300) and system (100) for controlling audio output. The method includes the steps of inputting (312) an audio signal and a voltage level signal, measuring (314) the audio signal and the voltage level signal, mapping (316) the audio signal against at least one table (134) of predetermined corresponding gain targets (138) and selecting (318) at least one gain target for the audio signal. The mapping step and the selecting step are based at least in part on the measurement of the voltage level signal and the measurement of the audio signal. The method also includes the step of applying (320) the gain target to the audio signal.

Abstract: The invention provides a cellular telephone or other communications device with intelligence to manage speakerphone operation to more nearly approximate normal conversation, even when using a one-way only transmission mode. The microphone path and speaker path may be continuously monitored using dual voice activity detectors to assess the energy and other characteristics of each channel, and switch between one or the other depending on dynamic criteria. In noisy environments, a hangtime may be applied before permitting switching to avoid premature dropouts. Other criteria used to trigger the seizure of the channel may be adjusted, such as to eliminate a lower threshold below which the speaker path switches out automatically.

Abstract: A communication device (200) for dual mode muting operation includes a user interface (208) and a controller (202). The controller (202) is programmed to: in response to a first user input to the user interface (208), enable a first muting mode; and in response to a second user input to the user interface (208), disable the first muting mode and enable a second muting mode. The first muting mode may be one of a full muting and a concealed muting, while the second muting mode may be the other one of a full muting and a concealed muting.