CEDAR RAPIDS — A 32-year-old Cedar Rapids man, who received probation for a high-speed chase that he bragged about as “fun” and attempted to elude again in March, is heading to prison.

Sixth Judicial District Judge Lars Anderson on Friday revoked probation for Travis McDermott on the eluding charge from June 9, 2019, and sentenced him to five years in prison.

McDermott was convicted Tuesday for attempting to elude in March and was sentenced to 90 days in jail, which was run concurrently to the five-year prison sentence.

First Assistant Linn County Attorney Nick Maybanks told the judge that McDermott has a “significant violent history,” including assaults, assault on a peace officer, domestic assault, interference with official acts and child endangerment with bodily injury.

He continues to assault others and “show blatant disregard for authority figures,” the prosecutor noted.

In the eluding incident from last June, McDermott “risked lives” in a southwest neighborhood leading police on chase that began on when police saw his vehicle speeding on Rockford Road SW and run a stop sign at Eighth Avenue SW, Maybanks said Friday. McDermott drove 107 mph in a 30 mph zone and drove the wrong way on a one-way street at Third Street and Wilson Avenue SW.

McDermott ran into a pile of dirt at a dead end, jumped out of his car and led officers on a foot chase, Maybanks said. He wouldn’t stop, and officers used a Taser to subdue him.

McDermott was laughing when police arrested him, saying “how much fun” he had and appearing to be under the influence of drugs or alcohol, Maybanks said.

McDermott demanded a speedy trial, but the officer who conducted the sobriety test wasn’t available for trial. A plea was offered, and the drunken driving charge was dropped.

Maybanks also pointed out McDermott wouldn’t cooperate with the probation office on a presentencing report, which was ordered by a judge. He picked up an assault charge last November and was convicted before his eluding sentencing in January.

Maybanks said after McDermott received probation, he didn’t show up at the probation office for his appointment, didn’t get a substance abuse test as ordered and reported an invalid address to community corrections.

McDermott also has a pending charge in Dubuque County for driving while barred March 3, according to court documents.

Comments: (319) 398-8318; trish.mehaffey@thegazette.com

Back in late 2016, we introduced Speech Recognition at Auphonic. This allows our users to create transcripts of their recordings, and more usefully, this means podcasts become searchable.
Now we integrated two more speech recognition engines: Amazon Transcribe and Speechmatics. Whilst integrating these services, we also took the opportunity to develop a complete new Transcription Editor:

Screenshot of our Transcript Editor with word confidence highlighting and the edit bar.
Try out the Transcript Editor Examples yourself!

The new Auphonic Transcript Editor is included directly in our HTML transcript output file, displays word confidence values to instantly see which sections should be checked manually, supports direct audio playback, HTML/PDF/WebVTT export and allows you to share the editor with someone else for further editing.

The new services, Amazon Transcribe and Speechmatics, offer transcription quality improvements compared to our other integrated speech recognition services.
They also return word confidence values, timestamps and some punctuation, which is exported to our output files.

The Auphonic Transcript Editor

With the integration of the two new services offering improved recognition quality and word timestamps alongside confidence scores, we realized that we could leverage these improvements to give our users easy-to-use transcription editing.
Therefore we developed a new, open source transcript editor, which is embedded directly in our HTML output file and has been designed to make checking and editing transcripts as easy as possible.

• Edit the transcription directly in the HTML document.
• Show/hide word confidence, to instantly see which sections should be checked manually (if you use Amazon Transcribe or Speechmatics as speech recognition engine).
• Listen to audio playback of specific words directly in the HTML editor.
• Share the transcript editor with others: as the editor is embedded directly in the HTML file (no external dependencies), you can just send the HTML file to some else to manually check the automatically generated transcription.
• Export the edited transcript to HTML, PDF or WebVTT.
• Completely useable on all mobile devices and desktop browsers.

Examples: Try Out the Transcript Editor

Here are two examples of the new transcript editor, taken from our speech recognition audio examples page:

1. Singletrack Transcript Editor Example

Singletrack speech recognition example from the first 10 minutes of Common Sense 309 by Dan Carlin. Speechmatics was used as speech recognition engine without any keywords or further manual editing.

2. Multitrack Transcript Editor Example

A multitrack automatic speech recognition transcript example from the first 20 minutes of TV Eye on Marvel - Luke Cage S1E1. Amazon Transcribe was used as speech recognition engine without any further manual editing.
As this is a multitrack production, the transcript includes exact speaker names as well (try to edit them!).

Transcript Editing

By clicking the Edit Transcript button, a dashed box appears around the text. This indicates that the text is now freely editable on this page. Your changes can be saved by using one of the export options (see below).
If you make a mistake whilst editing, you can simply use the undo/redo function of the browser to undo or redo your changes.

When working with multitrack productions, another helpful feature is the ability to change all speaker names at once throughout the whole transcript just by editing one speaker. Simply click on an instance of a speaker title and change it to the appropriate name, this name will then appear throughout the whole transcript.

Word Confidence Highlighting

Word confidence values are shown visually in the transcript editor, highlighted in shades of red (see screenshot above). The shade of red is dependent on the actual word confidence value: The darker the red, the lower the confidence value. This means you can instantly see which sections you should check/re-work manually to increase the accuracy.

Once you have edited the highlighted text, it will be set to white again, so it’s easy to see which sections still require editing.
Use the button Add/Remove Highlighting to disable/enable word confidence highlighting.

NOTE: Word confidence values are only available in Amazon Transcribe or Speechmatics, not if you use our other integrated speech recognition services!

Audio Playback

The button Activate/Stop Play-on-click allows you to hear the audio playback of the section you click on (by clicking directly on the word in the transcript editor).
This is helpful in allowing you to check the accuracy of certain words by being able to listen to them directly whilst editing, without having to go back and try to find that section within your audio file.

If you use an External Service in your production to export the resulting audio file, we will automatically use the exported file in the transcript editor.
Otherwise we will use the output file generated by Auphonic. Please note that this file is password protected for the current Auphonic user and will be deleted in 21 days.

If no audio file is available in the transcript editor, or cannot be played because of the password protection, you will see the button Add Audio File to add a new audio file for playback.

Export Formats, Save/Share Transcript Editor

Save/Share Editor

The Save Editor button stores the whole transcript editor with all its current changes into a new HTML file. Use this button to save your changes for further editing or if you want to share your transcript with someone else for manual corrections (as the editor is embedded directly in the HTML file without any external dependencies).

Export HTML / Export PDF / Export WebVTT

Use one of these buttons to export the edited transcript to HTML (for WordPress, Word, etc.), to PDF (via the browser print function) or to WebVTT (so that the edited transcript can be used as subtitles or imported in web audio players of the Podlove Publisher or Podigee).
Every export format is rendered directly in the browser, no server needed.

Amazon Transcribe

The first of the two new services, Amazon Transcribe, offers accurate transcriptions in English and Spanish at low costs, including keywords, word confidence, timestamps, and punctuation.

UPDATE 2019:
Amazon Transcribe offers more languages now - please see Amazon Transcribe Features!

Pricing

The free tier offers 60 minutes of free usage a month for 12 months. After that, it is billed monthly at a rate of $0.0004 per second ($1.44/h).
More information is available at Amazon Transcribe Pricing.

Custom Vocabulary (Keywords) Support

Custom Vocabulary (called Keywords in Auphonic) gives you the ability to expand and customize the speech recognition vocabulary, specific to your case (i.e. product names, domain-specific terminology, or names of individuals).
The same feature is also available in the Google Cloud Speech API.

Timestamps, Word Confidence, and Punctuation

Amazon Transcribe returns a timestamp and confidence value for each word so that you can easily locate the audio in the original recording by searching for the text.
It also adds some punctuation, which is combined with our own punctuation and formatting automatically.

The high-quality (especially in combination with keywords) and low costs of Amazon Transcribe make it attractive, despite only currently supporting two languages.
However, the processing time of Amazon Transcribe is much slower compared to all our other integrated services!

Try it yourself:
Connect your Auphonic account with Amazon Transcribe at our External Services Page.

Speechmatics

Speechmatics offers accurate transcriptions in many languages including word confidence values, timestamps, and punctuation.

Many Languages

Speechmatics’ clear advantage is the sheer number of languages it supports (all major European and some Asiatic languages).
It also has a Global English feature, which supports different English accents during transcription.

Timestamps, Word Confidence, and Punctuation

Like Amazon, Speechmatics creates timestamps, word confidence values, and punctuation.

Pricing

Speechmatics is the most expensive speech recognition service at Auphonic.
Pricing starts at £0.06 per minute of audio and can be purchased in blocks of £10 or £100. This equates to a starting rate of about $4.78/h. Reduced rate of £0.05 per minute ($3.98/h) are available if purchasing £1,000 blocks.
They offer significant discounts for users requiring higher volumes. At this further reduced price point it is a similar cost to the Google Speech API (or lower). If you process a lot of content, you should contact them directly at sales@speechmatics.com and say that you wish to use it with Auphonic.
More information is available at Speechmatics Pricing.

Speechmatics offers high-quality transcripts in many languages. But these features do come at a price, it is the most expensive speech recognition services at Auphonic.

Unfortunately, their existing Custom Dictionary (keywords) feature, which would further improve the results, is not available in the Speechmatics API yet.

Try it yourself:
Connect your Auphonic account with Speechmatics at our External Services Page.

What do you think?

Any feedback about the new speech recognition services, especially about the recognition quality in various languages, is highly appreciated.

We would also like to hear any comments you have on the transcript editor particularly - is there anything missing, or anything that could be implemented better?
Please let us know!

Until recently, Amazon Transcribe supported speech recognition in English and Spanish only.
Now they included French, Italian and Portuguese as well - and a few other languages (including German) are in private beta.

Update March 2019:
Now Amazon Transcribe supports German and Korean as well.

The Auphonic Audio Inspector on the status page of a finished Multitrack Production including speech recognition.
Please click on the screenshot to see it in full resolution!

Amazon Transcribe is integrated as speech recognition engine within Auphonic and offers accurate transcriptions (compared to other services) at low costs, including keywords / custom vocabulary support, word confidence, timestamps, and punctuation.
See the following AWS blog post and video for more information about recent Amazon Transcribe developments: Transcribe speech in three new languages: French, Italian, and Brazilian Portuguese.

Amazon Transcribe is also a perfect fit if you want to use our Transcript Editor because you will be able to see word timestamps and confidence values to instantly check which section/words should be corrected manually to increase the transcription accuracy:

Screenshot of our Transcript Editor with word confidence highlighting and the edit bar.

These features are also available if you use Speechmatics, but unfortunately not in our other integrated speech recognition services.

About Speech Recognition within Auphonic

Auphonic has built a layer on top of a few external speech recognition services to make audio searchable:
Our classifiers generate metadata during the analysis of an audio signal (music segments, silence, multiple speakers, etc.) to divide the audio file into small and meaningful segments, which are processed by the speech recognition engine. The results from all segments are then combined, and meaningful timestamps, simple punctuation and structuring are added to the resulting text.

To learn more about speech recognition within Auphonic, take a look at our Speech Recognition and Transcript Editor help pages or listen to our Speech Recognition Audio Examples.

A comparison table of our integrated services (price, quality, languages, speed, features, etc.) can be found here: Speech Recognition Services Comparison.

Conclusion

We hope that Amazon and others will continue to add new languages, to get accurate and inexpensive automatic speech recognition in many languages.

Don't hesitate to contact us if you have any questions or feedback about speech recognition or our transcript editor!

Even light exercise can counter the damage of stroke in survivors, a new study suggests.

Although the performance of today's speech recognition systems is impressive, the experience for many is still one of errors, corrections, frustration and abandoning speech in favour of alternative interaction methods. We take a closer look at speech and find out why speech recognition is so difficult.

This paper describes a dataset and protocols for evaluating continuous speech separation algorithms. Most prior studies on speech separation use pre-segmented signals of artificially mixed speech utterances which are mostly emph{fully} overlapped, and the algorithms are evaluated based on signal-to-distortion ratio or similar performance metrics. However, in natural conversations, a speech signal is continuous, containing both overlapped and overlap-free components. In addition, the signal-based metrics have very weak correlations with automatic speech recognition (ASR) accuracy. We think that not only does this make it hard to assess the practical relevance of the tested algorithms, it also hinders researchers from developing systems that can be readily applied to real scenarios. In this paper, we define continuous speech separation (CSS) as a task of generating a set of non-overlapped speech signals from a extit{continuous} audio stream that contains multiple utterances that are emph{partially} overlapped by a varying degree. A new real recorded dataset, called LibriCSS, is derived from LibriSpeech by concatenating the corpus utterances to simulate a conversation and capturing the audio replays with far-field microphones. A Kaldi-based ASR evaluation protocol is also established by using a well-trained multi-conditional acoustic model. By using this dataset, several aspects of a recently proposed speaker-independent CSS algorithm are investigated. The dataset and evaluation scripts are available to facilitate the research in this direction.

We present the Perceptimatic English Benchmark, an open experimental benchmark for evaluating quantitative models of speech perception in English. The benchmark consists of ABX stimuli along with the responses of 91 American English-speaking listeners. The stimuli test discrimination of a large number of English and French phonemic contrasts. They are extracted directly from corpora of read speech, making them appropriate for evaluating statistical acoustic models (such as those used in automatic speech recognition) trained on typical speech data sets. We show that phone discrimination is correlated with several types of models, and give recommendations for researchers seeking easily calculated norms of acoustic distance on experimental stimuli. We show that DeepSpeech, a standard English speech recognizer, is more specialized on English phoneme discrimination than English listeners, and is poorly correlated with their behaviour, even though it yields a low error on the decision task given to humans.

We propose Cotatron, a transcription-guided speech encoder for speaker-independent linguistic representation. Cotatron is based on the multispeaker TTS architecture and can be trained with conventional TTS datasets. We train a voice conversion system to reconstruct speech with Cotatron features, which is similar to the previous methods based on Phonetic Posteriorgram (PPG). By training and evaluating our system with 108 speakers from the VCTK dataset, we outperform the previous method in terms of both naturalness and speaker similarity. Our system can also convert speech from speakers that are unseen during training, and utilize ASR to automate the transcription with minimal reduction of the performance. Audio samples are available at https://mindslab-ai.github.io/cotatron, and the code with a pre-trained model will be made available soon.

Convolutional neural networks (CNN) have shown promising results for end-to-end speech recognition, albeit still behind other state-of-the-art methods in performance. In this paper, we study how to bridge this gap and go beyond with a novel CNN-RNN-transducer architecture, which we call ContextNet. ContextNet features a fully convolutional encoder that incorporates global context information into convolution layers by adding squeeze-and-excitation modules. In addition, we propose a simple scaling method that scales the widths of ContextNet that achieves good trade-off between computation and accuracy. We demonstrate that on the widely used LibriSpeech benchmark, ContextNet achieves a word error rate (WER) of 2.1\%/4.6\% without external language model (LM), 1.9\%/4.1\% with LM and 2.9\%/7.0\% with only 10M parameters on the clean/noisy LibriSpeech test sets. This compares to the previous best published system of 2.0\%/4.6\% with LM and 3.9\%/11.3\% with 20M parameters. The superiority of the proposed ContextNet model is also verified on a much larger internal dataset.

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The latest issue of the Inlander is hitting newsstands today. Find it at your local grocery store and hundreds of other locations; use this map to find a pickup point near you.…

Methods and apparatus for output of high-bandwidth debug data/traces in electronic devices using embedded high-speed debug port(s). Debug data is received from multiple blocks and buffered in a buffer. The buffer's output is operatively coupled to one or more high-speed serial I/O interfaces via muxing logic during debug test operations. The buffered data is encoded as serialized data and sent over the one or more high-speed serial I/O interfaces to a logic device that receives serialized data and de-serializes it to generate parallel debug data that is provided to a debugger. The buffer may be configured as a bandwidth-adapting buffer that facilitates transfer of debug data that is received at a variable combined data rate outbound via the one or more high-speed serial I/O interfaces at a data rate corresponding to the bandwidth of the serial I/O interfaces.

A double data rate memory physical interface having self checking loopback logic on-chip is disclosed. Disposed on the chip is a first linear feedback shift register, which is capable of generating a set of test data values that comprise at least two data bits. Also disposed on the chip is a second linear feedback shift register. The second linear feedback shift register is capable of generating a set of expected data values that match the test data values. Further, an internal loopback error check element is disposed on the chip. The internal loopback error check element is used to compare the set of expected data values with the set of test data values.

A barrel shifter uses a sign magnitude to 2's complement converter to generate decoder signals for its cascaded multiplexer selectors. The sign input receives the shift direction and the magnitude input receives the shift amount. The sign magnitude to 2's complement converter computes an output result as a 2's complement of the shift amount using the shift direction as a sign input, assigns a first portion (most significant bit half) of the output result to a first decoder signal, and assigns a second portion (least significant bit half) of the output result to a second decoder signal. This encoding scheme allows the decoder circuits to be relatively simple, for example, 3-to-8 decoders for an implementation adapted to shift a 64-bit operand value rather than the 4-to-9 decoder required in a conventional barrel shifter, leading to faster operation, less area, and reduced power consumption.

A method of adjusting a speed of a mobile machine is provided. Image data of a location is collected where currently generated sensor data and previously generated sensor data indicate a discontinuity in sensor data. The image data is analyzed to determine if a non-motion blur score for the image data is above a threshold value. Then, a speed of the mobile machine is adjusted based on a determination that the non-motion blur score is above the threshold value.

Apparatus and methods are provided for using automatic speech recognition to analyze a voice interaction and verify compliance of an agent reading a script to a client during the voice interaction. In one aspect of the invention, a communications system includes a user interface, a communications network, and a call center having an automatic speech recognition component. In other aspects of the invention, a script compliance method includes the steps of conducting a voice interaction between an agent and a client and evaluating the voice interaction with an automatic speech recognition component adapted to analyze the voice interaction and determine whether the agent has adequately followed the script. In yet still further aspects of the invention, the duration of a given interaction can be analyzed, either apart from or in combination with the script compliance analysis above, to seek to identify instances of agent non-compliance, of fraud, or of quality-analysis issues.

A device may include a physical phenomenon detector. The physical phenomenon detector may detect a physical phenomenon related to the device. In response to detecting the physical phenomenon, the device may record audio data that includes speech. The speech may be transcribed with a speech recognition engine. The speech recognition engine may be included in the device, or may be included with a remote computing device with which the device may communicate.

Some embodiments of the inventive subject matter include a method for detecting speech loss and supplying appropriate recollection data to the user. Such embodiments include detecting a speech stream from a user, converting the speech stream to text, storing the text, detecting an interruption to the speech stream, wherein the interruption to the speech stream indicates speech loss by the user, searching a catalog using the text as a search parameter to find relevant catalog data and, presenting the relevant catalog data to remind the user about the speech stream.

A speech recognition system, method of recognizing speech and a computer program product therefor. A client device identified with a context for an associated user selectively streams audio to a provider computer, e.g., a cloud computer. Speech recognition receives streaming audio, maps utterances to specific textual candidates and determines a likelihood of a correct match for each mapped textual candidate. A context model selectively winnows candidate to resolve recognition ambiguity according to context whenever multiple textual candidates are recognized as potential matches for the same mapped utterance. Matches are used to update the context model, which may be used for multiple users in the same context.

A speech recognition method including the steps of receiving a speech input from a known speaker of a sequence of observations and determining the likelihood of a sequence of words arising from the sequence of observations using an acoustic model. The acoustic model has a plurality of model parameters describing probability distributions which relate a word or part thereof to an observation and has been trained using first training data and adapted using second training data to said speaker. The speech recognition method also determines the likelihood of a sequence of observations occurring in a given language using a language model and combines the likelihoods determined by the acoustic model and the language model and outputs a sequence of words identified from said speech input signal. The acoustic model is context based for the speaker, the context based information being contained in the model using a plurality of decision trees and the structure of the decision trees is based on second training data.

A method and apparatus for using an ad hoc peer-to-peer distributed message bus is provided. The method may include determining, by a local bus node, using a first power level communication protocol, that a remote bus node is available, obtaining, by the local bus node, connection information from the remote bus node using a second power level communication protocol, wherein the connection information comprises connection information for one or more remote endpoints associated with the remote bus node, and generating one or more local virtual endpoints, wherein each of the one or more local virtual endpoints corresponds to each of the one or more remote endpoints, and wherein the remote endpoint is described with reference to a well-known name, unique to the remote endpoint.

When a speech sound of at least a predetermined sound pressure is externally input while a time measurement is not being performed, a time measuring circuit starts a time measurement responsive to a signal from a speech detector. When another speech sound of at least a predetermined sound pressure is externally input while a time measurement is being performed by the time measuring circuit, a measurement time measured by the time measuring circuit at this moment is stored in a time information memory. After a predetermined time has elapsed, if a speech recognition circuit recognizes that the externally input speech sound is a "stop" command, the time measurement operation performed by the time measuring circuit is stopped, and the time information stored in the time information memory is read out and displayed as measurement time information on a display unit.

The present invention pertains to a process for carrying out a high-speed stop in a Fischer-Tropsch process which comprises providing a feed to a fixed bed reactor comprising a Fischer-Tropsch catalyst, the reactor being at reaction temperature and pressure, and withdrawing an effluent from the reactor, wherein the high-speed stop is effected by blocking provision of feed to the reactor and simultaneously blocking the withdrawal of effluent from the reactor.

In the present invention, one or both of an active wheel or a passive wheel is composed of an eccentric transmission wheel and is combined with a synchronous transmission belt for forming an eccentric wheel transmission wheel series, so that when the feet input forces at different angles from the treadle shafts of the treadles to an active wheel shaft combined on the active wheel through cranks, the active wheel forms different transmission speed ratios relative to the passive wheel according to the treadle angle, and random inward packing is performed to the transmission belt (100) of the engage end of the eccentric passive wheel (413) during the transmission for stabling the operation.

The present invention is structured by using one or both of an active wheel and a passive wheel being composed of an eccentric transmission wheel and being combined with a synchronous transmission belt for forming an eccentric wheel transmission wheel series, so that in the reciprocal treadle performed by the human's feet, when the feet input forces at different angles from the treadle shafts of the treadles to an active wheel shaft combined on the active wheel through cranks, the active wheel forms different transmission speed ratios relative to the passive wheel according to the treadle angle.

Methods and systems for decoding a signal include compensating for impairments in a received signal using at least carrier phase estimation, where residual phase error remains after compensation; calculating symbol log-likelihood ratios (LLRs) for symbols in the compensated signal using Monte Carlo integration; demapping the symbols in the compensated signal using the symbol LLRs and extrinsic information from signal decoding to produce one or more estimated codewords; and decoding each estimated codeword with a decoder that generates a decoded codeword and extrinsic information.

A system and method of high-speed microscopy using a two-photon microscope with spectral resolution. The microscope is operable to provide two- to five-dimensional fluorescence images of samples, including two or three spatial dimensions, a spectral dimension (for fluorescence emission), and a temporal dimension (on a scale of less than approximately one second). Two-dimensional (spatial) images with a complete wavelength spectrum are generated from a single scan of a sample. The microscope may include one of a multi-beam point scanning microscope, a single beam line scanning microscope, and a multi-beam line scanning microscope. The line scans may be formed using one or more of curved mirrors and lenses. The multiple beams may be formed using one of a grating, an array of lenses, and a beam splitter.

A wireless media player and a related system and methodology are disclosed. One aspect of the wireless media player system pertains to a virtual connector system, apparatus, and method for the automatic establishment of wireless connectivity with other electronic devices. In one embodiment, the media player device employs the use of integrated Radio Frequency Identification (RFID) technology to exchange communication settings, media capability, and other parameters with an external device that also has integrated RFID technology. The automatic exchange of settings and other information via a proximity-based RFID data exchange allows a media player to quickly establish a secure communication link with another device via a commonly supported wireless protocol such as Ultra Wideband (UWB) or Bluetooth. Another aspect of the media player system pertains to a method of using the captured media capability of the connecting device to customize certain menu options and software parameters in the media player.

A transmission includes an input member, an output member, four planetary gear sets, and a plurality of torque transmitting mechanisms that are selectively engageable to establish at least ten forward speed ratios and at least one reverse speed ratio between the input member and the output member. The transmission further includes one or more locking mechanisms that engage one or more of the plurality of torque transmitting mechanisms during a start/stop event.

A method, an apparatus, and a computer program product for wireless communication are provided in which a UE is equipped to generate an expression associated with a wireless device for use in a peer discovery signal, generate an expression set from the expression, a first temporal frequency, and a first time duration, and transmit the generated expression set to a base station to allow the base station to broadcast each of the expressions included in the expression set at the first temporal frequency over the first time duration. Another method, apparatus, and computer program product for wireless communication are provided in which a base station is equipped to receive an expression set from a wireless device, process the received expression set to determine each of one or more instances of an expression, and transmit each of the one or more instances of the expression.

A variable speed power converter controls the speed of a load in a material handling system as a function of the torque required to move the load. While the power converter is running, the torque being produced in the motor is determined. The power converter then determines the maximum rotational speed of the motor as a function of the torque currently being produced and of the torque-speed curve of the motor. The power converter then commands the motor to rotate at this maximum rotational speed. The power converter periodically monitors the torque being produced and adjusts the maximum rotational speed of the motor throughout the run.

A variable speed trigger mechanism that allows a user to reverse a direction of a motor and supply variable amounts of power to the motor using a single trigger mechanism. In a first motion, the user can actuate the reversing module to change the direction of the motor coupled to the trigger mechanism. In a second motion, the user can actuate the same trigger and apply variable amounts of power to the motor.

A circuit for speed monitoring of an electric motor comprises a circuit for generating a time-frame signal, a circuit for receiving a first signal from a chopper driver circuit designed to drive the electric motor, a circuit for detecting chopper pulses in the first signal, a pulse counter, and a circuit for at least one of outputting and evaluating a state of the pulse counter, after the inactive state of the time-frame has been indicated. The time-frame signal indicates when a time-frame of predefined length changes from an inactive state to an active state and indicates when the time-frame changes back from the active state to the inactive state. The pulse counter is designed to count the detected chopper pulses while the active state is indicated by the circuit for generating the time-frame signal.

A computer-implemented fan control method includes measuring a temperature within a computer system and dynamically selecting a fan speed step in response to the temperature received, wherein the fan speed step is selected from a fan speed table defining a finite number of fan speed steps each having an associated fan speed. A fan is operated at the dynamically selected fan speed step, wherein the fan is positioned to drive air through the computer system where the temperature is being measured. The fan output variation is measured over a prescribed time interval and the fan speed table is automatically modified to change the fan speeds associated with each fan speed step, wherein the fan speeds are changed as a function of the measured fan output variation while continuing to drive the fan.

A circuit includes a comparator having input terminals configured to be coupled across a drive transistor adapted to drive a phase of a motor. The comparator senses a drive current of the motor phase, said sensed drive current represented by a periodic signal whose period is indicative of motor speed. A motor speed calculation circuit receives the periodic signal and processes the periodic signal to determine a speed of the motor.

A motor driving apparatus is applied to a fan and motor mechanism and a voltage supply unit. The motor driving apparatus includes a motor driving unit, a voltage division resistor, a first resistor, a first switch unit, a second resistor, a second switch unit, a third resistor, a third switch unit, a transistor switch, and a pulse width modulation unit. The first switch unit, the second switch unit, and the third switch unit are configured to select the rotational speed upper limitation of the fan and motor mechanism for suppressing noise.

A rotation speed control circuit is disclosed. The rotation speed control circuit includes a temperature-controlled voltage duty generator, a pulse-width signal duty generator, a multiplier and a rotation speed signal generator. The temperature-controlled voltage duty generator converts temperature-controlled voltage to digital temperature-controlled voltage and executes linear interpolation operation according to a first setting data so as to output temperature-controlled voltage duty signal. The pulse-width signal duty generator coverts pulse-width input signal to a digital pulse-width input signal and executes linear interpolation operation according to a second setting data so as to output a pulse-width duty signal. The temperature-controlled voltage duty signal and the pulse-width duty signal are executed for multiplication by the multiplier so as to output mixing-duty signal. The rotation speed generator receives the mixing-duty signal and a third setting data, and executes a minimum output duty operation so as to output a pulse-width output signal.

A system for determining motor speed of a brush DC motor in an apparatus, including a first filter for receiving a substantially DC component of the motor current and parameters corresponding to the brush DC motor, for calculating a speed estimate thereof; an adaptive bandpass filter having a center frequency corresponding to the speed estimate of the first filter, for receiving the motor current and substantially isolating a periodic current fluctuation thereof; a block for determining a frequency of the periodic current fluctuation, the current fluctuation corresponding to motor speed of the brush DC motor. The adaptive bandpass filter uses debounce filtering to reduce rapid filter passband switching, and a run-in period prior to passband switching to obviate transient effects of filter switching.

A transceiver includes a transmitter and receiver that form a series current path between two power-supply nodes. Powering both the transmitter and receiver with the same supply current saves power. The transmitter functions as a resistive load for the receiver, and thus performs useful work with power that would otherwise be dissipated as waste heat.

System and method for testing a high speed data path without generating a high speed bit clock, includes selecting a first high speed data path from a plurality of data paths for testing. Coherent clock data patterns are driven on one or more of remaining data paths of the plurality of data paths, wherein the coherent clock data patterns are in coherence with a low speed base clock. The first high speed data path is sampled by the coherent clock data patterns to generate a sampled first high speed data path, which is then tested at a speed of the low speed base clock.

A method of complementing a spoken text. The method including receiving text data representative of a natural language text, receiving effect control data including at least one effect control record, each effect control record being associated with a respective location in the natural language text, receiving a stream of audio data, analyzing the stream of audio data for natural language utterances that correlate with the natural language text at a respective one of the locations, and outputting, in response to a determination by the analyzing that a natural language utterance in the stream of audio data correlates with a respective one of the locations, at least one effect control signal based on the effect control record associated with the respective location.

Guard elements (31) are provided for the corner connectors (16 through 19) of a heald shaft (10) of a shedding unit, the guard elements covering the open space formed between the two legs (26, 27) and thus providing a grip protection.

An electrical machine includes a stator having a main armature winding, an exciter field winding, and a transformer primary winding. A rotor is operatively connected to rotate relative to the stator, wherein the rotor includes an exciter armature winding operatively connected to the exciter armature winding for field excitation therebetween, a main field winding operatively connected to the main armature winding for field excitation therebetween, and a transformer secondary winding operatively connected to the transformer primary winding to form a rotating transformer. A generator control unit is operatively connected to the main armature winding, exciter field winding, and transformer primary winding to control the main armature and exciter field windings based on excitation in the primary winding received from the transformer secondary winding.

Some embodiments relate to a method of controlling speed of a variable speed generator. The method includes detecting a load of the variable speed generator and determining a target speed for the variable speed generator based on the load supplied by the variable speed generator. The method further includes using a controller to adjust the speed of the variable speed generator based on the target speed. The method may further include correcting the target speed by calculating a correction factor that corrects the target speed based on a voltage produced by the variable speed generator.

A method for managing a drilling operation, including generating, by a first sensor and a second sensor of a bottom hole assembly (BHA), a first time based data log and a second time based data log, respectively, representing a borehole parameter along a drilling trajectory, determining, by a computer processor of the BHA and during the drilling operation, a time shift by comparing the first time based data log and the second time based data log, where offsetting the first and second time based data logs by the time shift maximizes a correlation factor of the first and second time based data logs, and determining, within a pre-determined time period from generating the first and second time based data logs, a drilling speed based on the time shift and a pre-determined distance between the first sensor and the second sensor.

Embodiments of digital high-speed bi-phase modulator and method for bi-phase modulation are generally described herein. In some embodiments, the digital high-speed bi-phase modulator comprises a high-speed digital divider, a high-speed digital multiplexer, and matched signal paths provided between the divider and the multiplexer. The high-speed digital divider is configured to receive a carrier signal and generate complementary output signals. The high-speed digital multiplexer is configured to switch between the complementary output signals and generate a bi-phase modulated output at a carrier frequency (fc) modulated with a bi-phase code. The bi-phase code may be provided to control inputs of the multiplexer.

Embodiments include systems and methods for accurately controlling gain of a high-speed variable-gain amplifier (VGA) without adversely impacting bandwidth performance. Embodiments include a VGA with a variable resistor, for which resistance is a function of a control level. A gain calibration system controls the control level by using a gain control feedback subsystem to sample outputs of a duplicate VGA, which includes a duplicate variable resistor. The sampled duplicate outputs are compared to a target gain generated by a reference generator. The control level can be fed back to control the gain of the duplicate VGA until the target gain is reached. The control level can also be fed to the actual VGA to control its gain. By performing gain control on the duplicate VGA without interfering with the output signal path of the actual VGA, the actual VGA's gain can be accurately controlled without impacting its bandwidth.

The present invention is a serial to parallel data conversion method and device where new serial data are stored within a first n-bit register prior to presentation at an n-bit parallel output. Subsequently, additional data are stored within a second n-bit register while the data stored within the first register are presented at the parallel output. Data storage and data presentation are thereafter alternated, thereby eliminating the problem of setup time seen in prior art.

A device for coupling tilting railroad cars onto a high-speed railroad train comprises a coupler and a shock absorber held in a shock absorber frame, each joined by a coupling pin via a coupling joint. The frame includes a front inner surface with a front end wall opening and a cylindrical surface extending to a semi-circular concave inner surface. The joint includes a rounded convex surface engaged with a semi-circular concave surface, a circumferential surface of a short cylinder receiving the rear portion of the coupler, and an insertion hole which is provided in a direction perpendicular to the axis of the short cylinder. The frame has a removal hole and an air removal pin of the coupling pin facing each other at a position to rotate at a predetermined angle in a circumferential direction in the assembled state, and at a position coaxial with the coupling pin in the assembled state.

A direct drive drawworks (100) has a permanent magnet motor (40) with a first set of windings (250) and a second set of windings (252), a shaft (41) extending from the permanent magnet motor (40) such that the permanent magnet motor directly rotates the shaft (41), a drum (43) connected to the shaft (41) away from the permanent magnet motor (40) such that the rotation of the shaft (41) causes a corresponding rotation of the drum (43), and a switch cooperative with the first set of windings and the second set of windings so as to cause the sets of windings to be selectively connected in parallel or in series.