Relevant bibliographies by topics / Restaurace signálu

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  1. Journal articles
  2. Dissertations / Theses
  3. Conference papers

Academic literature on the topic 'Restaurace signálu'

Author: Grafiati
Published: 4 June 2021
Last updated: 13 February 2022

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Journal articles on the topic "Restaurace signálu"

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Lin, Michael S., Amit Sharma, and Yuxia Ouyang. "Role of Signals In Consumers’ Economic Valuation of Restaurant Choices." Journal of Hospitality & Tourism Research 44, no. 7 (May 31, 2020): 1080–100. http://dx.doi.org/10.1177/1096348020926238.

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While businesses provide “signals” related to products/services to alleviate information asymmetry and facilitate consumers’ economic valuation of their offerings, it is not always clear which of these signals is more effective. This study investigated the comparative effectiveness of signals on influencing consumers’ choices and their willingness to pay (WTP) for a restaurant meal. Results of an online stated choice experiment (N =328) suggest that local independent ownership, local sourcing, and food quality significantly influenced individuals’ choices and increased WTP for a meal. Higher food quality signal in a local and independently owned restaurant was associated with a 40% WTP price premium than in a national chain restaurant. Findings of this study contribute to our understanding of signaling factors’ relative influence on an individual’s decision. The study also offers management implications for restaurant owners on how to effectively align signals to target potential consumers.
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Dorman, Michael F., Sarah Natale, and Louise Loiselle. "Speech Understanding and Sound Source Localization by Cochlear Implant Listeners Using a Pinna-Effect Imitating Microphone and an Adaptive Beamformer." Journal of the American Academy of Audiology 29, no. 03 (March 2018): 197–205. http://dx.doi.org/10.3766/jaaa.16126.

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AbstractSentence understanding scores for patients with cochlear implants (CIs) when tested in quiet are relatively high. However, sentence understanding scores for patients with CIs plummet with the addition of noise.To assess, for patients with CIs (MED-EL), (1) the value to speech understanding of two new, noise-reducing microphone settings and (2) the effect of the microphone settings on sound source localization.Single-subject, repeated measures design. For tests of speech understanding, repeated measures on (1) number of CIs (one, two), (2) microphone type (omni, natural, adaptive beamformer), and (3) type of noise (restaurant, cocktail party). For sound source localization, repeated measures on type of signal (low-pass [LP], high-pass [HP], broadband noise).Ten listeners, ranging in age from 48 to 83 yr (mean = 57 yr), participated in this prospective study.Speech understanding was assessed in two noise environments using monaural and bilateral CIs fit with three microphone types. Sound source localization was assessed using three microphone types.In Experiment 1, sentence understanding scores (in terms of percent words correct) were obtained in quiet and in noise. For each patient, noise was first added to the signal to drive performance off of the ceiling in the bilateral CI-omni microphone condition. The other conditions were then administered at that signal-to-noise ratio in quasi-random order. In Experiment 2, sound source localization accuracy was assessed for three signal types using a 13-loudspeaker array over a 180° arc. The dependent measure was root-mean-score error.Both the natural and adaptive microphone settings significantly improved speech understanding in the two noise environments. The magnitude of the improvement varied between 16 and 19 percentage points for tests conducted in the restaurant environment and between 19 and 36 percentage points for tests conducted in the cocktail party environment. In the restaurant and cocktail party environments, both the natural and adaptive settings, when implemented on a single CI, allowed scores that were as good as, or better, than scores in the bilateral omni test condition. Sound source localization accuracy was unaltered by either the natural or adaptive settings for LP, HP, or wideband noise stimuli.The data support the use of the natural microphone setting as a default setting. The natural setting (1) provides better speech understanding in noise than the omni setting, (2) does not impair sound source localization, and (3) retains low-frequency sensitivity to signals from the rear. Moreover, bilateral CIs equipped with adaptive beamforming technology can engender speech understanding scores in noise that fall only a little short of scores for a single CI in quiet.
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Dorman, Michael F., and Sarah Cook Natale. "AutoAdaptive: A Noise Level–Sensitive Beamformer for MED EL Cochlear Implant Patients." Journal of the American Academy of Audiology 30, no. 08 (September 2019): 731–34. http://dx.doi.org/10.3766/jaaa.18030.

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AbstractWhen cochlear implant (CI) listeners use a directional microphone or beamformer system to improve speech understanding in noise, the gain in understanding for speech presented from the front of the listener coexists with a decrease in speech understanding from the back. One way to maximize the usefulness of these systems is to keep a microphone in the omnidirectional mode in low noise and then switch to directional mode in high noise.The purpose of this experiment was to assess the levels of speech understanding in noise allowed by a new signal processing algorithm for MED EL CIs, AutoAdaptive, which operates in the manner described previously.Seven listeners fit with bilateral CIs were tested in a simulation of a crowded restaurant with speech presented from the front and from the back at three noise levels, 45, 55, and 65 dB SPL.The listeners were seated in the middle of an array of eight loudspeakers. Sentences from the AzBio sentence lists were presented from loudspeakers at 0 or 180° azimuth. Restaurant noise at 45, 55, and 65 dB SPL was presented from all eight loudspeakers. The speech understanding scores (words correct) were subjected to a two-factor (speaker location and noise level), repeated measures, analysis of variance with posttests.The analysis of variance showed a main effect for level and location and a significant interaction. Posttests showed that speech understanding scores from front and back loudspeakers did not differ significantly at the 45- and 55-dB noise levels but did differ significantly at the 65-dB noise level—with increased scores for signals from the front and decreased scores for signals from the back.The AutoAdaptive feature provides omnidirectional benefit at low noise levels, i.e., similar levels of speech understanding for talkers in front of, and in back of, a listener and beamformer benefit at higher noise levels, i.e., increased speech understanding for signals from in front. The automatic switching feature will be of value to the many patients who prefer not to manually switch programs on their CIs.
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Drew, Jordan A., and W. Owen Brimijoin. "Negative impacts from latency masked by noise in simulated beamforming." PLOS ONE 16, no. 7 (July 1, 2021): e0254119. http://dx.doi.org/10.1371/journal.pone.0254119.

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Those experiencing hearing loss face severe challenges in perceiving speech in noisy situations such as a busy restaurant or cafe. There are many factors contributing to this deficit including decreased audibility, reduced frequency resolution, and decline in temporal synchrony across the auditory system. Some hearing assistive devices implement beamforming in which multiple microphones are used in combination to attenuate surrounding noise while the target speaker is left unattenuated. In increasingly challenging auditory environments, more complex beamforming algorithms are required, which increases the processing time needed to provide a useful signal-to-noise ratio of the target speech. This study investigated whether the benefits from signal enhancement from beamforming are outweighed by the negative impacts on perception from an increase in latency between the direct acoustic signal and the digitally enhanced signal. The hypothesis for this study is that an increase in latency between the two identical speech signals would decrease intelligibility of the speech signal. Using 3 gain / latency pairs from a beamforming simulation previously completed in lab, perceptual thresholds of SNR from a simulated use case were obtained from normal hearing participants. No significant differences were detected between the 3 conditions. When presented with 2 copies of the same speech signal presented at varying gain / latency pairs in a noisy environment, any negative intelligibility effects from latency are masked by the noise. These results allow for more lenient restrictions for limiting processing delays in hearing assistive devices.
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Nageswara Rao, S., K. Jaya Sankar, and C. D. Naidu. "An Improved Bi-Level Thresholding Based Uncertainty Evaluation for Speech Enhancement in Non-Stationary Noises." International Journal of Engineering & Technology 7, no. 2.24 (April 25, 2018): 436. http://dx.doi.org/10.14419/ijet.v7i2.24.12130.

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This paper proposes a new speech enhancement framework to improve the quality of speeches recorded under adverse acoustic environments based on the speech presence uncertainty. Since the uncertainty evaluation gives a more and clear discrimination about the speech and noise, this paper proposes a new uncertainty evaluation mechanism as a preprocessing mechanism to the noise suppression methods. This mechanism relates with energies of a noisy speech signal and classifies the speech segments and noise segments more perfectly. In addition to the quality enhancement, this approach also reduces the unnecessary computational burden over the speech processing system. Extensive simulations are carried out over the speech signals with different types of non-stationary noises like babble noise, exhibition noise, restaurant noise and train station noises and the performance is measured with the performance metrics namely the Output SNR, AvgSegSNR, PESQ and COMP. The comparative analysis of proposed approach over the conventional approaches shows an outstanding performance in all environments.
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Herranz, Luis, Shuqiang Jiang, and Ruihan Xu. "Modeling Restaurant Context for Food Recognition." IEEE Transactions on Multimedia 19, no. 2 (February 2017): 430–40. http://dx.doi.org/10.1109/tmm.2016.2614861.

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Sahni, Navdeep S., and Harikesh S. Nair. "Does Advertising Serve as a Signal? Evidence from a Field Experiment in Mobile Search." Review of Economic Studies 87, no. 3 (October 8, 2019): 1529–64. http://dx.doi.org/10.1093/restud/rdz053.

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Abstract We develop a field experiment that assesses whether advertising can serve as a signal that enhances consumers’ evaluations of advertised goods. We implement the experiment on a mobile search platform that provides listings and reviews for an archetypal experience good, restaurants. In collaboration with the platform, we randomize about 200,000 users in 13 Asian cities into exposure of ads for about 600+ local restaurants. Within the exposure group, we randomly vary the disclosure to the consumer of whether a restaurant’s listing is a paid-ad. This enables isolating the effect on outcomes of a user knowing that a listing is sponsored—a pure signalling effect. We find that this disclosure increases calls to the restaurant by 77%, holding fixed all other attributes of the ad. The disclosure effect is higher when the consumer uses the platform away from his typical city of search, when the uncertainty about restaurant quality is larger, and for restaurants that have received fewer ratings in the past. On the supply side, newer, higher rated and more popular restaurants are found to advertise more on the platform; and ratings of those that advertised during the experiment are found to be higher two years later. Taken together, we interpret these results as consistent with a signalling equilibrium in which ads serve as implicit signals that enhance the appeal of the advertised restaurants to consumers. Both consumers and advertisers seem to benefit from the signalling. Consumers shift choices towards restaurants that are better rated (at baseline) in the disclosure group compared to the no disclosure group, and advertisers gain from the improved outcomes induced by disclosure.
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Vo Nhu, Thanh. "Development of restaurant serving robot using line following approach." Journal of Science and Technology Issue on Information and Communications Technology 17, no. 12.1 (December 31, 2019): 1. http://dx.doi.org/10.31130/jst-ud2019-087.

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As the era of industrial revolution 4.0 is coming, most of robots are developed to replace the human works and services. However, in Vietnam, putting robots in service industry such as waiter robot serving in restaurants is still very limited. This study presents the initial development of restaurant serving robot using line following approach. The robot is designed by referring to the survey on the size and appearance requirements at a restaurant. Robot is programmed to come to a specific table by mapping data. Based on the line reading algorithm implemented on microcontroller PIC18F4550, the robot follows the line marked on the floor to move to the desired table position and returns to the service counter after completing the task. There are ultrasonic sensors on the robot that can detect the obstacle to help the robot stop and output warning signal. The robot also can output some simple voices to customers in English, Korean and Vietnamese such as "Welcome", “Please enjoy your food"...
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Võ Như, Thành. "Development of restaurant serving robot using line following approach." Journal of Science and Technology Issue on Information and Communications Technology 17, no. 12.1 (December 31, 2019): 1. http://dx.doi.org/10.31130/jst-ud2019-087e.

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As the era of industrial revolution 4.0 is coming, most of robots are developed to replace the human works and services. However, in Vietnam, putting robots in service industry such as waiter robot serving in restaurants is still very limited. This study presents the initial development of restaurant serving robot using line following approach. The robot is designed by referring to the survey on the size and appearance requirements at a restaurant. Robot is programmed to come to a specific table by mapping data. Based on the line reading algorithm implemented on microcontroller PIC18F4550, the robot follows the line marked on the floor to move to the desired table position and returns to the service counter after completing the task. There are ultrasonic sensors on the robot that can detect the obstacle to help the robot stop and output warning signal. The robot also can output some simple voices to customers in English, Korean and Vietnamese such as "Welcome", “Please enjoy your food"...
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Dorman, Michael F., Sarah Cook Natale, and Smita Agrawal. "The Value of Unilateral CIs, CI-CROS and Bilateral CIs, with and without Beamformer Microphones, for Speech Understanding in a Simulation of a Restaurant Environment." Audiology and Neurotology 23, no. 5 (2018): 270–76. http://dx.doi.org/10.1159/000493844.

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Objective: Our primary aim was to determine, in a simulation of a crowded restaurant, the value to speech understanding of (i) a unilateral cochlear implant (CI), (ii) a CI plus CROS (contralateral routing of signals) aid system and (iii) bilateral CIs when tested with and without beamforming microphones. Design: The listeners were 7 CI listeners who had used bilateral CIs for an average of 9 years. The listeners were tested with three device configurations (bilateral CI, unilateral CI + CROS, and unilateral CI), two signal processing conditions (without and with beamformers) and with speech either from +90°, –90°, or from the front. Speech understanding scores for the TIMIT sentences were obtained in the 8-loudspeaker R-SPACETM test environment – an environment which simulates listening in a crowded restaurant. Results: In the unilateral condition, speech understanding, relative to speech directed to the CI ear, fell by 17% when speech was from the front and fell 28% when speech was to the side opposite the CI. These deficits were overcome with both CI-CROS and bilateral CIs, and scores for the two devices did not differ significantly for any location of speech input. Beamformer microphones improved speech understanding for speech from the front and depressed speech understanding for speech from the sides for all device configurations. Patients with bilateral CIs and beamformers achieved slightly, but significantly, higher scores for speech from the front than patients with CI-CROS and beamformers. Conclusions: CI-CROS is a valuable addition to the hardware options available to patients fit with a single CI. For patients fit with bilateral CIs, bilateral beamformers are a valuable addition in the condition of speech coming from in front of the listener. The small differences in performance in the CI-CROS and bilateral CI conditions suggest that patient preference for bilateral CIs is based largely on factors other than speech understanding in noise.
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Dissertations / Theses on the topic "Restaurace signálu"

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Hájek, Vojtěch. "Restaurace signálu s omezenou okamžitou hodnotou pro vícekanálový audio signál." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-401987.

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This master’s thesis deals with the restoration of clipped multichannel audio signals based on sparse representations. First, a general theory of clipping and theory of sparse representations of audio signals is described. A short overview of existing restoration methods is part of this thesis as well. Subsequently, two declipping algorithms are introduced and are also implemented in the Matlab environment as a part of the thesis. The first one, SPADE, is considered a state- of-the-art method for mono audio signals declipping and the second one, CASCADE, which is derived from SPADE, is designed for the restoration of multichannel signals. In the last part of the thesis, both algorithms are tested and the results are compared using the objective measures SDR and PEAQ, and also using the subjective listening test MUSHRA.
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Beňo, Tomáš. "Restaurace signálu s omezenou okamžitou hodnotou s použitím psychoakustického modelu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-401986.

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The master's thesis deals with the restoration of audio signals that have been damaged by clipping. Used methods are based on sparse representations of signals. The introduction of the thesis explains the issue of clipping and mentions the list of already existing methods that solve declipping, which are followed by the thesis. In the next chapter, the necessary theory of sparse representations and the proximal algorithms is described, including specific representatives from the category of convex optimization problems. The thesis contains declipping algorithm implemented in Matlab software environment. Chosen method for solving the task uses the Condat algorithm or Generic proximal algorithm for convex optimization and solves minimization of sum of three convex functions. The result of the thesis is five versions of algorithm and three of them have implemented psychoacoustic model for results improvement. For each version has been found optimal setting of parameters. The restoration quality results are evaluated using objective measurements like SDR and PEMO-Q and also using subjective listening test.
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Záviška, Pavel. "Restaurace audiosignálů založená na řídkých reprezentacích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2017. http://www.nusl.cz/ntk/nusl-317028.

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This Master's Thesis deals with the issue of audio clipping and the application of sparse represenations model for the task of declipping. First, a general theory of clipping is described, followed by a brief overview of existing methods and a description of the general theory concerning sparse representations of signals and bases, respectively frames. Subsequently, two methods solving declipping problem based on sparse representations are intruduced. The first method uses the Generic proximal algorithm for convex optimization, the second one uses the Douglas-Rachford algorithm. The above mentioned methods have been programmed in the Matlab environment. The results of the declipping methods are evaluated according to SNR, PEMO-Q and also by subjective listening tests.
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Mach, Václav. "Doplňování chybějících vzorků v audio signálu." Doctoral thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2016. http://www.nusl.cz/ntk/nusl-240542.

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V oblasti zpracování signálů se v současné době čím dál více využívají tzv. řídké reprezentace signálů, tzn. že daný signál je možné vyjádřit přesně či velmi dobře aproximovat lineární kombinací velmi malého počtu vektorů ze zvoleného reprezentačního systému. Tato práce se zabývá využitím řídkých reprezentací pro rekonstrukci poškozených zvukových záznamů, ať už historických nebo nově vzniklých. Především historické zvukové nahrávky trpí zarušením jako praskání nebo šum. Krátkodobé poškození zvukových nahrávek bylo doposud řešeno interpolačními technikami, zejména pomocí autoregresního modelování. V nedávné době byl představen algoritmus s názvem Audio Inpainting, který řeší doplňování chybějících vzorků ve zvukovém signálu pomocí řídkých reprezentací. Zmíněný algoritmus využívá tzv. hladové algoritmy pro řešení optimalizačních úloh. Cílem této práce je porovnání dosavadních interpolačních metod s technikou Audio Inpaintingu. Navíc, k řešení optimalizačních úloh jsou využívány algoritmy založené na l1-relaxaci, a to jak ve formě analyzujícího, tak i syntetizujícího modelu. Především se jedná o proximální algoritmy. Tyto algoritmy pracují jak s jednotlivými koeficienty samostatně, tak s koeficienty v závislosti na jejich okolí, tzv. strukturovaná řídkost. Strukturovaná řídkost je dále využita taky pro odšumování zvukových nahrávek. Jednotlivé algoritmy jsou v praktické části zhodnoceny z hlediska nastavení parametrů pro optimální poměr rekonstrukce vs. výpočetní čas. Všechny algoritmy popsané v práci jsou na praktických příkladech porovnány pomocí objektivních metod odstupu signálu od šumu (SNR) a PEMO-Q. Na závěr je úspěšnost rekonstrukce poškozených zvukových signálů vyhodnocena.
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Skyva, Pavel. "Moderní metody potlačování šumu v audiosignálu založené na fázi." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2019. http://www.nusl.cz/ntk/nusl-401981.

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The thesis deals with modern methods of audio denoising. Reconstruction of the audiosignal is primarly based on utilization of phase information of signals and phase derivatives. Denoising methods also use sparse signal representations. In thesis is described the way of searching sparse coefficients using proximal Condat algorithm and following computation of reconstructed signal using this coefficients. The reconstruction algorithms are implemented in the MATLAB software with toolbox LTFAT included. Results of the reconstruction are compared using objective evaluation method Signal-to-Noise Ratio (SNR) and also by subjective evaluation.
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Šiška, Jakub. "Restaurace zvukových signálů poškozených kvantizací." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2020. http://www.nusl.cz/ntk/nusl-413249.

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This master’s thesis deals with the restoration of audio signals damaged by quantization. The theoretical part starts with a description of quantization and dequantization in general, few existing methods of dequantization of audio signals and theory of sparse representations of signals are also presented. The next part introduces algorithms suitable for dequantization, specifically Douglas–Rachford, Chambolle–Pock, SPADEQ and implementation of these algorithms in MATLAB application in the next chapter. In the last part of this thesis, testing of reconstructed signals using the algorithms takes place and results are evaluated by objective measures SDR, PEMO-Q, PEAQ and subjective listening test MUSHRA.
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Conference papers on the topic "Restaurace signálu"

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Imperiale, Anthony S., and Allen E. Milewski. "User-Centered Development of a Dynamic Restaurant Menu Information Appliance." In Imaging and Signal Processing in Health Care and Technology. Calgary,AB,Canada: ACTAPRESS, 2012. http://dx.doi.org/10.2316/p.2012.772-023.

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Elifoglu, Murat, and Tunga Gungor. "A restaurant recommendation system for Turkish based on user conversations." In 2018 26th Signal Processing and Communications Applications Conference (SIU). IEEE, 2018. http://dx.doi.org/10.1109/siu.2018.8404153.

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Kim, Seokhwan, and Rafael E. Banchs. "R-cube: A dialogue agent for restaurant recommendation and reservation." In 2014 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA). IEEE, 2014. http://dx.doi.org/10.1109/apsipa.2014.7041732.

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Park, Taehyeun, and Walid Saad. "Kolkata paise restaurant game for resource allocation in the Internet of Things." In 2017 51st Asilomar Conference on Signals, Systems, and Computers. IEEE, 2017. http://dx.doi.org/10.1109/acssc.2017.8335666.

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Yang, Chao, Lei Xie, and Xiangzeng Zhou. "Unsupervised broadcast news story segmentation using distance dependent Chinese restaurant processes." In ICASSP 2014 - 2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2014. http://dx.doi.org/10.1109/icassp.2014.6854365.

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Dasgupta, Soham, Madhurupa Samaddar, CN Yogalakshmi, K. Vijayan, and Subhiksha. "Autonomous Restaurant Serving System using Image Detection and Voiceprint Detection in Android Application." In 2020 International Conference on Communication and Signal Processing (ICCSP). IEEE, 2020. http://dx.doi.org/10.1109/iccsp48568.2020.9182314.

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Takehara, Masanori, Hiroya Nojiri, Satoshi Tamura, Satoru Hayamizu, and Takeshi Kurata. "Analysis of customer communication by employee in restaurant and lead time estimation." In 2014 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA). IEEE, 2014. http://dx.doi.org/10.1109/apsipa.2014.7041701.

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