This document provides an overview of using audio test and measurement instruments to evaluate broadcast facilities. It discusses using multitone signals to test audio quality over the air and evaluate new equipment in real time. Case studies are presented on using audio analyzers to diagnose issues with a Bluetooth car kit by measuring speech quality scores and analyzing frequency response characteristics. The document emphasizes the importance of objective measurement for ensuring broadcast audio quality.
This document describes a LabVIEW project to create a chromatic tuner. The tuner allows users to select an input, plays the corresponding note, and displays if the played note is sharp or flat using lights and a tuning needle. It was tested by playing guitar strings and verifying the tuner accurately identified in-tune and out-of-tune notes. The tuner provides three methods for users to tune themselves: lights indicating pitch, a tuning dial, and playing the target note.
A Distributed System for Recognizing Home Automation Commands and Distress Ca...a3labdsp
The document describes a distributed system that recognizes home automation commands and distress calls in Italian. It consists of two units: a Local Multimedia Control Unit that recognizes commands/calls and manages communication, and a Central Management Unit that integrates home services and handles emergencies. The system uses acoustic echo cancellation and speech recognition to understand commands even in noisy environments. An evaluation of the system showed it achieved over 90% accuracy on headset microphone data and over 50% on distant microphone data.
Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate. It provides extremely accurate estimates of speech parameters, and is relatively efficient for computation.
This document discusses linear predictive coding (LPC) methods and horn noise detection. It begins with an introduction to speech coders and speech production modeling. It then covers the basic principles of LPC analysis, including the autocorrelation and covariance methods. It discusses solving the LPC equations and using LPC residue to detect horn noise by comparing the residue of speech, silence and known horn noise samples. The document provides results of adding speech and horn noise signals and detecting the horn noise. It concludes by listing references on speech coding algorithms, LPC, and speech processing.
Digital signal processing involves representing and processing signals in the form of discrete numeric values. It has various applications including radar, biomedical monitoring, speech recognition, communications, image processing, and multimedia. Key aspects of digital signal processing implementation are analog to digital conversion, digital processing, and digital to analog conversion. Limitations include information loss due to sampling, aliasing effects, limited frequency resolution, and quantization error. However, digital signal processing provides advantages such as reprogrammability, accuracy control, easy storage and transport of signals, and ability to implement sophisticated algorithms.
The document discusses speech processing and vocoding. It begins by defining speech and how it is produced, including voiced and unvoiced sounds. It then describes the human speech production system and various speech coding techniques like waveform coding, vocoding, and analysis-by-synthesis coding. Finally, it provides details on the G.729 speech codec, including its operations, process flow, specifications, and how it achieves speech compression to 8 kbps from the original 128 kbps.
DSP_FOEHU - Lec 13 - Digital Signal Processing Applications IAmr E. Mohamed
This document provides an overview of digital signal processing applications including digital spectrum analysis, speech processing, and radar. It discusses different types of digital spectrum analyzers including filter bank, swept, and FFT analyzers. It also covers topics related to speech processing like the anatomy of speech production, speech perception, voiced and unvoiced sounds, and phonemes. Common speech coding techniques are introduced such as vocoding, ADPCM, LPC, and CELP coding. Radar applications of DSP are also briefly mentioned.
This document describes a LabVIEW project to create a chromatic tuner. The tuner allows users to select an input, plays the corresponding note, and displays if the played note is sharp or flat using lights and a tuning needle. It was tested by playing guitar strings and verifying the tuner accurately identified in-tune and out-of-tune notes. The tuner provides three methods for users to tune themselves: lights indicating pitch, a tuning dial, and playing the target note.
A Distributed System for Recognizing Home Automation Commands and Distress Ca...a3labdsp
The document describes a distributed system that recognizes home automation commands and distress calls in Italian. It consists of two units: a Local Multimedia Control Unit that recognizes commands/calls and manages communication, and a Central Management Unit that integrates home services and handles emergencies. The system uses acoustic echo cancellation and speech recognition to understand commands even in noisy environments. An evaluation of the system showed it achieved over 90% accuracy on headset microphone data and over 50% on distant microphone data.
Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate. It provides extremely accurate estimates of speech parameters, and is relatively efficient for computation.
This document discusses linear predictive coding (LPC) methods and horn noise detection. It begins with an introduction to speech coders and speech production modeling. It then covers the basic principles of LPC analysis, including the autocorrelation and covariance methods. It discusses solving the LPC equations and using LPC residue to detect horn noise by comparing the residue of speech, silence and known horn noise samples. The document provides results of adding speech and horn noise signals and detecting the horn noise. It concludes by listing references on speech coding algorithms, LPC, and speech processing.
Digital signal processing involves representing and processing signals in the form of discrete numeric values. It has various applications including radar, biomedical monitoring, speech recognition, communications, image processing, and multimedia. Key aspects of digital signal processing implementation are analog to digital conversion, digital processing, and digital to analog conversion. Limitations include information loss due to sampling, aliasing effects, limited frequency resolution, and quantization error. However, digital signal processing provides advantages such as reprogrammability, accuracy control, easy storage and transport of signals, and ability to implement sophisticated algorithms.
The document discusses speech processing and vocoding. It begins by defining speech and how it is produced, including voiced and unvoiced sounds. It then describes the human speech production system and various speech coding techniques like waveform coding, vocoding, and analysis-by-synthesis coding. Finally, it provides details on the G.729 speech codec, including its operations, process flow, specifications, and how it achieves speech compression to 8 kbps from the original 128 kbps.
DSP_FOEHU - Lec 13 - Digital Signal Processing Applications IAmr E. Mohamed
This document provides an overview of digital signal processing applications including digital spectrum analysis, speech processing, and radar. It discusses different types of digital spectrum analyzers including filter bank, swept, and FFT analyzers. It also covers topics related to speech processing like the anatomy of speech production, speech perception, voiced and unvoiced sounds, and phonemes. Common speech coding techniques are introduced such as vocoding, ADPCM, LPC, and CELP coding. Radar applications of DSP are also briefly mentioned.
Nicholas Ambrosio proposes to design a chromatic tuner using an Arduino and FFT algorithm. The tuner will detect frequencies within a concert B-flat chromatic scale. It will have an LED display and screen to provide live FFT feedback. So far, Nicholas has designed a pre-amplifier and offset circuit. He is using an FFT library to analyze frequencies but needs to improve the resolution. Increasing the sample rate and FFT size could help distinguish frequencies more accurately.
Speech coding techniques are used to represent human speech in a digital form for applications like mobile communication and voice over IP. The main components of a speech coding system are speech encoding and decoding. Various coding techniques are used including waveform coding techniques like PCM and ADPCM, and source coding techniques like linear predictive coding (LPC) and vocoding. The aim is to enhance speech quality at a particular bitrate or minimize the bitrate at a given quality level, while considering factors like computational complexity, coding delay, and robustness to different speakers.
Digital speech within 125 hz bandwidth (DS-125)Michael Lebo
This document summarizes a software project called Digital Speech within 125 Hz Bandwidth (DS-125). The project aims to transmit live voice over long distances using a low bandwidth of 125 Hz by breaking down voice into short audio clips and unique identification codes. It discusses how voice is sampled and converted into binary digits sent at 125 Hz, then reconstructed at the receiving end into synthetic voice. The project seeks to address voice transmission for applications like communication with astronauts on Mars by using a low bandwidth to overcome signal degradation issues.
The document describes the Digital Tuner Project conducted at TU Berlin in the summer of 2012. The project involved building a digital guitar tuner board that uses a microphone to input sound signals, performs digital signal processing to analyze the frequency spectrum, and outputs the identified guitar note and its tune status via LED lights. The document provides details on the hardware components of the tuner board, the software used, the digital signal processing concepts applied, and an overview of the microcontroller implementation.
Interactive voice conversion for augmented speech productionNU_I_TODALAB
This document discusses recent progress in interactive voice conversion techniques for augmenting speech production. It begins by explaining the physical limitations of normal speech production and how voice conversion can augment speech by controlling more information. It then discusses how interactive voice conversion allows for quick response times, better controllability through real-time feedback, and understanding user intent from multimodal behavior signals. Recent advances discussed include low-latency voice conversion networks, controllable waveform generation respecting the source-filter model of speech, and expression control using signals like arm movements. The goal is to develop cooperatively augmented speech that can help users with lost speech abilities.
This document discusses various techniques for speech coding used in digital communication systems. It covers fundamental concepts like sampling theory, quantization, predictive coding, and linear predictive coding (LPC). It then describes specific speech codecs including PCM, ADPCM, CELP, LD-CELP, ACELP, and LPC vocoders. It discusses characteristics of speech coding like being lossy and metrics like SNR and MOS. Finally, it provides details on widely used standards like G.711, G.729, G.723.1, and GSM.
Speech Analysis and synthesis using VocoderIJTET Journal
Abstract— In this paper, I proposed a speech analysis and synthesis using a vocoder. Voice conversion systems do not create new speech signals, but just transform existing one. The proposed speech vocoding is different from speech coding. To analyze the speech signal and represent it with less number of bits, so that bandwidth efficiency can be increased. The Synthesis of speech signal from the received bits of information. In this paper three aspects of analysis have been discussed: pitch refinement, spectral envelope estimation and maximum voiced frequency estimation. A Quasi-harmonic analysis model can be used to implement a pitch refinement algorithm which improves the accuracy of the spectral estimation. Harmonic plus noise model to reconstruct the speech signal from parameter. Finally to achieve the highest possible resynthesis quality using the lowest possible number of bits to transmit the speech signal. Future work aims at incorporating the phase information into the analysis and modeling process and also synthesis these three aspects in different pitch period.
Digital audio was created in the late 1960s when Dr. Thomas Stockham began experimenting with digital tape recording using analog to digital converters. The key aspects of digital audio are:
1) Analog audio is converted to digital form through analog to digital conversion which samples the analog signal at regular intervals determined by the sample rate.
2) Higher sample rates and bit depths produce more accurate digital representations of the original analog signal but result in larger file sizes.
3) Quantization error, in the form of quantization noise, occurs when sample values are rounded to binary numbers during digitization and can be reduced by dithering and increasing bit depth.
Analogue sound storage has advantages like using less bandwidth and providing a more accurate representation of sound. However, it is susceptible to white noise distortion and quality degradation. Digital sound storage allows for easy editing and access without degrading quality. However, digital sounds can lose quality and are reliant on computer storage and functionality. Both analogue and digital signals are converted from sound waves to electrical signals and vice versa using devices like ADCs and DACs to store, transmit, and playback sound files.
The document provides information about the Waves L2 audio processor software. The L2 combines advanced peak limiting, level maximization, and Increased Digital Resolution (IDR) processing. IDR uses dithering and noise shaping to maximize digital resolution when reducing bit depth. The L2 allows processing at 48-bit precision and outputting at 24-bit for archiving. It includes Auto Release Control and offers various dither types, noise shaping curves, and output bit depths for different applications.
This document discusses speech coding techniques. It introduces speech processing which includes compression, manipulation, storage, transfer and reconstruction of speech. It explains that speech coding is needed for applications like mobile communication, voice over internet protocol, satellite broadcasting and PSTN networks. The document outlines various attributes of speech coders like low bit-rate, high speech quality and low computational complexity. It also describes different types of speech coders like waveform coders, source coders, time domain coders and frequency domain coders. Objective and subjective measures for evaluating performance of speech coders are discussed.
This document introduces digital audio by explaining the difference between analog and digital signals. It describes key variables that affect audio sampling including sampling rate, bit depth, and number of channels. Higher sampling rates, bit depths, and more channels captured result in higher quality audio files but also larger file sizes. The optimal balance of these variables must be determined based on the intended use and quality needed for the audio.
This document proposes a method called Digital Speech within 125 Hz Bandwidth (DS-125) to synthesize live voice using extremely short audio clips within a narrow bandwidth. It describes how sound detection could identify sounds in 0.008 second intervals and assign a binary code to each one. Distortion is addressed by overlapping audio clips. A "Lebo code" is introduced to represent speech and timing with sequences of ones and zeros mapped to pre-recorded audio clips. The document calls for further work to fully implement the system and test the proof-of-concept.
[NUGU CONFERENCE 2019] 트랙 A-2 : NUGU call 적용 기술 및 서비스 소개NUGU developers
NUGU call is a hands-free calling platform that allows connections anywhere through NUGU Touch Points. It supports multi-device connections under one account and has voice UX features like initiating and ending calls through voice commands. Voice quality is maintained through standards for loudness, frequency response, and other factors. Real-time communication uses internet protocols and signaling for call setup, media exchange, and termination. Upcoming features will expand NUGU call to support video calls and intelligent contextual commands.
1) Digital communication systems can transmit signals over long distances using regenerators that recover the original data sequence at each segment. This allows for much lower error rates than analog systems.
2) The maximum transmission rate through a channel is 2 times the channel bandwidth. More data can be sent by increasing the number of amplitude levels used to encode each pulse. However, noise places an ultimate limit on transmission rate as it increases bit error rates.
3) Shannon's channel capacity formula describes the maximum reliable transmission rate for a channel based on its bandwidth and signal-to-noise ratio. Rates below the capacity can achieve arbitrarily low error rates with sufficient coding.
DIGITAL SIGNAL PROCESWSING AND ITS APPLICATIONLokeshBanarse
Digital signal processing (DSP) involves using digital technology to process analog signals. It converts analog signals into digital data that can be manipulated and analyzed. DSP has applications in areas like audio processing, image processing, radar, and mobile phones. The key components of DSP systems are program memory, data memory, a compute engine, and input/output interfaces. DSP emerged in the 1960s and was initially used for applications like radar, sonar, and space exploration. It later expanded into commercial uses with the growth of personal computers and consumer electronics.
This proposal outlines the development of a multi-media neurofeedback system for therapeutic use. It would use a wireless EEG headset to measure brainwaves and transform them into sound and visual feedback through software. The software would analyze the brainwaves in real-time and use multiple channels of sound and 3D graphics to guide the user's mental state. Therapists could customize neurofeedback protocols and track user data online to monitor progress. The system aims to provide an engaging way to train attention and help treat various conditions through neurofeedback therapy.
The document discusses analogue and digital signals. It explains that analogue signals exist on a sliding scale while digital signals are either on or off. It notes that analogue signals are prone to noise but don't require complex equipment, while digital signals can eliminate noise through fast and clever electronics, though these are needed to process digital signals. It provides examples of analogue and digital devices and signals.
An analogue signal is a continuous signal that varies smoothly over a range of values, unlike a digital signal which has discrete steps. Some key advantages of analogue signals are their potentially infinite resolution and simpler processing requirements compared to digital signals. However, analogue signals are susceptible to noise that can distort the signal over time or distance. Examples of analogue signals and recordings include vinyl records, tape recordings, and analogue audio equipment.
This document is a curriculum vitae for Bodala Jagadeesh summarizing his skills and experience. He has 1.5 years of experience in designing data warehousing projects using Informatica and Oracle. He has created mappings to extract data from various sources, load data into databases, and create reusable objects. He also has experience with Tableau creating dashboards from various data sources according to client requirements. His technical skills include Informatica, Oracle, Postgres, SQL, and Tableau.
This document is a chapter from a student project on photovoltaic solar power plants. It includes an introduction to PV solar technology that discusses grid-connected and off-grid PV systems, solar cell types, conversion efficiency, and factors affecting PV performance. It also provides details on the major components of a PV plant such as electrical buildings, inverters, DC systems, modules and arrays. The appendices include examples of annual power generation and CO2 reduction from solar as well as a glossary of solar terms.
Nicholas Ambrosio proposes to design a chromatic tuner using an Arduino and FFT algorithm. The tuner will detect frequencies within a concert B-flat chromatic scale. It will have an LED display and screen to provide live FFT feedback. So far, Nicholas has designed a pre-amplifier and offset circuit. He is using an FFT library to analyze frequencies but needs to improve the resolution. Increasing the sample rate and FFT size could help distinguish frequencies more accurately.
Speech coding techniques are used to represent human speech in a digital form for applications like mobile communication and voice over IP. The main components of a speech coding system are speech encoding and decoding. Various coding techniques are used including waveform coding techniques like PCM and ADPCM, and source coding techniques like linear predictive coding (LPC) and vocoding. The aim is to enhance speech quality at a particular bitrate or minimize the bitrate at a given quality level, while considering factors like computational complexity, coding delay, and robustness to different speakers.
Digital speech within 125 hz bandwidth (DS-125)Michael Lebo
This document summarizes a software project called Digital Speech within 125 Hz Bandwidth (DS-125). The project aims to transmit live voice over long distances using a low bandwidth of 125 Hz by breaking down voice into short audio clips and unique identification codes. It discusses how voice is sampled and converted into binary digits sent at 125 Hz, then reconstructed at the receiving end into synthetic voice. The project seeks to address voice transmission for applications like communication with astronauts on Mars by using a low bandwidth to overcome signal degradation issues.
The document describes the Digital Tuner Project conducted at TU Berlin in the summer of 2012. The project involved building a digital guitar tuner board that uses a microphone to input sound signals, performs digital signal processing to analyze the frequency spectrum, and outputs the identified guitar note and its tune status via LED lights. The document provides details on the hardware components of the tuner board, the software used, the digital signal processing concepts applied, and an overview of the microcontroller implementation.
Interactive voice conversion for augmented speech productionNU_I_TODALAB
This document discusses recent progress in interactive voice conversion techniques for augmenting speech production. It begins by explaining the physical limitations of normal speech production and how voice conversion can augment speech by controlling more information. It then discusses how interactive voice conversion allows for quick response times, better controllability through real-time feedback, and understanding user intent from multimodal behavior signals. Recent advances discussed include low-latency voice conversion networks, controllable waveform generation respecting the source-filter model of speech, and expression control using signals like arm movements. The goal is to develop cooperatively augmented speech that can help users with lost speech abilities.
This document discusses various techniques for speech coding used in digital communication systems. It covers fundamental concepts like sampling theory, quantization, predictive coding, and linear predictive coding (LPC). It then describes specific speech codecs including PCM, ADPCM, CELP, LD-CELP, ACELP, and LPC vocoders. It discusses characteristics of speech coding like being lossy and metrics like SNR and MOS. Finally, it provides details on widely used standards like G.711, G.729, G.723.1, and GSM.
Speech Analysis and synthesis using VocoderIJTET Journal
Abstract— In this paper, I proposed a speech analysis and synthesis using a vocoder. Voice conversion systems do not create new speech signals, but just transform existing one. The proposed speech vocoding is different from speech coding. To analyze the speech signal and represent it with less number of bits, so that bandwidth efficiency can be increased. The Synthesis of speech signal from the received bits of information. In this paper three aspects of analysis have been discussed: pitch refinement, spectral envelope estimation and maximum voiced frequency estimation. A Quasi-harmonic analysis model can be used to implement a pitch refinement algorithm which improves the accuracy of the spectral estimation. Harmonic plus noise model to reconstruct the speech signal from parameter. Finally to achieve the highest possible resynthesis quality using the lowest possible number of bits to transmit the speech signal. Future work aims at incorporating the phase information into the analysis and modeling process and also synthesis these three aspects in different pitch period.
Digital audio was created in the late 1960s when Dr. Thomas Stockham began experimenting with digital tape recording using analog to digital converters. The key aspects of digital audio are:
1) Analog audio is converted to digital form through analog to digital conversion which samples the analog signal at regular intervals determined by the sample rate.
2) Higher sample rates and bit depths produce more accurate digital representations of the original analog signal but result in larger file sizes.
3) Quantization error, in the form of quantization noise, occurs when sample values are rounded to binary numbers during digitization and can be reduced by dithering and increasing bit depth.
Analogue sound storage has advantages like using less bandwidth and providing a more accurate representation of sound. However, it is susceptible to white noise distortion and quality degradation. Digital sound storage allows for easy editing and access without degrading quality. However, digital sounds can lose quality and are reliant on computer storage and functionality. Both analogue and digital signals are converted from sound waves to electrical signals and vice versa using devices like ADCs and DACs to store, transmit, and playback sound files.
The document provides information about the Waves L2 audio processor software. The L2 combines advanced peak limiting, level maximization, and Increased Digital Resolution (IDR) processing. IDR uses dithering and noise shaping to maximize digital resolution when reducing bit depth. The L2 allows processing at 48-bit precision and outputting at 24-bit for archiving. It includes Auto Release Control and offers various dither types, noise shaping curves, and output bit depths for different applications.
This document discusses speech coding techniques. It introduces speech processing which includes compression, manipulation, storage, transfer and reconstruction of speech. It explains that speech coding is needed for applications like mobile communication, voice over internet protocol, satellite broadcasting and PSTN networks. The document outlines various attributes of speech coders like low bit-rate, high speech quality and low computational complexity. It also describes different types of speech coders like waveform coders, source coders, time domain coders and frequency domain coders. Objective and subjective measures for evaluating performance of speech coders are discussed.
This document introduces digital audio by explaining the difference between analog and digital signals. It describes key variables that affect audio sampling including sampling rate, bit depth, and number of channels. Higher sampling rates, bit depths, and more channels captured result in higher quality audio files but also larger file sizes. The optimal balance of these variables must be determined based on the intended use and quality needed for the audio.
This document proposes a method called Digital Speech within 125 Hz Bandwidth (DS-125) to synthesize live voice using extremely short audio clips within a narrow bandwidth. It describes how sound detection could identify sounds in 0.008 second intervals and assign a binary code to each one. Distortion is addressed by overlapping audio clips. A "Lebo code" is introduced to represent speech and timing with sequences of ones and zeros mapped to pre-recorded audio clips. The document calls for further work to fully implement the system and test the proof-of-concept.
[NUGU CONFERENCE 2019] 트랙 A-2 : NUGU call 적용 기술 및 서비스 소개NUGU developers
NUGU call is a hands-free calling platform that allows connections anywhere through NUGU Touch Points. It supports multi-device connections under one account and has voice UX features like initiating and ending calls through voice commands. Voice quality is maintained through standards for loudness, frequency response, and other factors. Real-time communication uses internet protocols and signaling for call setup, media exchange, and termination. Upcoming features will expand NUGU call to support video calls and intelligent contextual commands.
1) Digital communication systems can transmit signals over long distances using regenerators that recover the original data sequence at each segment. This allows for much lower error rates than analog systems.
2) The maximum transmission rate through a channel is 2 times the channel bandwidth. More data can be sent by increasing the number of amplitude levels used to encode each pulse. However, noise places an ultimate limit on transmission rate as it increases bit error rates.
3) Shannon's channel capacity formula describes the maximum reliable transmission rate for a channel based on its bandwidth and signal-to-noise ratio. Rates below the capacity can achieve arbitrarily low error rates with sufficient coding.
DIGITAL SIGNAL PROCESWSING AND ITS APPLICATIONLokeshBanarse
Digital signal processing (DSP) involves using digital technology to process analog signals. It converts analog signals into digital data that can be manipulated and analyzed. DSP has applications in areas like audio processing, image processing, radar, and mobile phones. The key components of DSP systems are program memory, data memory, a compute engine, and input/output interfaces. DSP emerged in the 1960s and was initially used for applications like radar, sonar, and space exploration. It later expanded into commercial uses with the growth of personal computers and consumer electronics.
This proposal outlines the development of a multi-media neurofeedback system for therapeutic use. It would use a wireless EEG headset to measure brainwaves and transform them into sound and visual feedback through software. The software would analyze the brainwaves in real-time and use multiple channels of sound and 3D graphics to guide the user's mental state. Therapists could customize neurofeedback protocols and track user data online to monitor progress. The system aims to provide an engaging way to train attention and help treat various conditions through neurofeedback therapy.
The document discusses analogue and digital signals. It explains that analogue signals exist on a sliding scale while digital signals are either on or off. It notes that analogue signals are prone to noise but don't require complex equipment, while digital signals can eliminate noise through fast and clever electronics, though these are needed to process digital signals. It provides examples of analogue and digital devices and signals.
An analogue signal is a continuous signal that varies smoothly over a range of values, unlike a digital signal which has discrete steps. Some key advantages of analogue signals are their potentially infinite resolution and simpler processing requirements compared to digital signals. However, analogue signals are susceptible to noise that can distort the signal over time or distance. Examples of analogue signals and recordings include vinyl records, tape recordings, and analogue audio equipment.
This document is a curriculum vitae for Bodala Jagadeesh summarizing his skills and experience. He has 1.5 years of experience in designing data warehousing projects using Informatica and Oracle. He has created mappings to extract data from various sources, load data into databases, and create reusable objects. He also has experience with Tableau creating dashboards from various data sources according to client requirements. His technical skills include Informatica, Oracle, Postgres, SQL, and Tableau.
This document is a chapter from a student project on photovoltaic solar power plants. It includes an introduction to PV solar technology that discusses grid-connected and off-grid PV systems, solar cell types, conversion efficiency, and factors affecting PV performance. It also provides details on the major components of a PV plant such as electrical buildings, inverters, DC systems, modules and arrays. The appendices include examples of annual power generation and CO2 reduction from solar as well as a glossary of solar terms.
Nandan Lorekar is seeking a position that utilizes his 2.6 years of experience in roles such as computer operator, HR executive, and administrator. He has strong skills in Microsoft Office, problem solving, multi-tasking, and time management. His resume provides details on his work history and responsibilities in previous roles supporting HR, administration, and computer operations.
Fiama Di Wills targets young, modern, skin-conscious people between 18-35 years old. It segments the market based on age, lifestyle, and priorities. Fiama Di Wills positions itself as a high-quality, affordable soap that provides moisture and care for the skin, appealing to middle-class men and women seeking everyday skin care products. The gel soap bar contains conditioners for rich moisturization at an accessible price point.
Kundan Kumar Singh is a civil engineer with over 4 years of experience in construction field, quality control, and project management. He is looking for a new opportunity to utilize his skills and knowledge. He has experience in planning, monitoring, resource allocation, quality assurance, and supervision of projects. He also has technical skills in estimation, design, AutoCAD, and conducting lab tests. His most recent role was as Quality Control Engineer at Jindal SAW Limited.
The document outlines the professional approach and core values of integrity, respect, collaboration, empowerment and responsibility. It then lists the industries served, including mega construction projects, hospitality, healthcare, oil and gas, manufacturing, power and utility, engineering procurement and construction, mining, IT and telecommunications, and operation and maintenance. Key services provided are ensuring confidentiality, having a global reach, customer satisfaction, quality project delivery, development process support, and daily status updates.
This short document promotes the creation of Haiku Deck presentations on SlideShare and encourages the reader to get started making their own presentation. It contains three stock photo credits but no other text.
The document introduces a new smart briefcase called the S2 briefcase. It is an all-in-one briefcase that offers security, functionality and gadget handling features like biometric locking, GPS tracking, and charging ports. The briefcase aims to capture 25-30% of the luggage market share by providing best-in-class products and becoming a well-known brand with maximum consumer trust. A marketing plan is proposed to promote the S2 briefcase and increase awareness and sales.
The feedback was regarding a magazine cover and film poster design. The designs needed some adjustments to better communicate the intended messages. Specifically, the magazine cover photo did not clearly convey the article's topic and the film poster text was too small and busy. Suggestions were provided to select a more relevant photo and simplify the poster design.
CloudEngage Powering the Geo-Responsive WebTom Williams
Personalization should start with the first visit to your site, but it has to be fast and lightweight or it just won't get used.
CloudEngage geolocation offers super easy personalization with custom content based on location of each viewer and local weather. 91% of marketers can't be wrong.
This document discusses various film marketing techniques including the use of nonlinear narratives, fast cuts, and tense sounds in trailers. It also mentions establishing themes through imagery on posters, magazine covers, and careful use of mis-en-scene, lighting, fonts and camera work. The text also notes challenges in managing audience expectations and promoting films to succeed financially.
The document discusses the benefits of exercise for mental health. Regular physical activity can help reduce anxiety and depression and improve mood and cognitive function. Exercise causes chemical changes in the brain that may help protect against mental illness and improve symptoms.
Kundan Kumar Singh is seeking a position as a civil engineer with 3 years of experience in the construction industry. He has a Bachelor's degree in Civil Engineering and has worked on projects involving pile foundations, water treatment plants, overhead tanks, and pipelines. His responsibilities have included site supervision, quality control testing, schedule preparation, and liaising with clients and contractors. He has strengths in project management, technical skills, and safety and quality monitoring.
OneDrive is a free online file storage service that allows users to store and access files from any device. Files can be added to OneDrive from a computer by dragging them into the OneDrive folder, from a phone or tablet using the OneDrive app, or from any device using the OneDrive website. Once files are in OneDrive, they can be easily shared or collaborated on with others. OneDrive also integrates with Windows and Office programs to allow files to be accessed and edited from any device.
Tarek Adel Thabet is an experienced program manager with over 16 years of experience in telecommunications. He holds an MBA and a Master's Certificate in Project Management from George Washington University. Currently he is leading various packet core projects and upgrades for Nokia in Egypt.
This document provides strategies for choosing a major, including completing written exercises to reflect on interests and goals, asking the right questions about potential majors, researching majors through the university catalog and talking to current students, gaining experience through coursework, job shadowing or internships, and exploring careers through assessments at the Career Services office. It notes that about 1/3 of college students enter undeclared and over 50% change majors. The final steps are meeting with an advisor in the chosen major and completing a declaration of major form.
This document discusses age ratings and certifications for movies in the US and UK. It provides examples of how two movies, Copycat and Single White Female, received R ratings from the MPAA for violence, language and sexuality. It also explains that the BBFC and MPAA help determine age ratings and content descriptions to help parents decide if a film is suitable for their children. The document concludes that the creator's movie would be rated 15 by the BBFC due to its strong violence, language, sexual content and references to sex and violence.
How do you Cool Your Brew? Cool Brewing Fermentation CoolerCool-brewing
This short document shares 4 photos from different photographers to inspire creativity. It encourages the viewer to create their own presentation on SlideShare using Haiku Deck, a tool for making simple slideshows. A brief call to action is given to get started making a presentation.
La topología tiene sus orígenes en los trabajos de Euler, Cantor y Möbius. Fue Poincaré quien publicó en 1895 el trabajo Análisis Situs que es considerado como el punto decisivo en el desarrollo de la topología. Hausdorff creó en 1914 la teoría de espacios abstractos usando la noción de vecindario y estableció las bases de la topología conjuntista como disciplina matemática propia.
Measuring Loudness Levels § When we measure loudness levels, we need to account for frequency sensitivity § This is done by applying a weighting filter to audio § For several yeas the A-weighting filter was commonly used § Other weighting curves (B and C for example) are also defined § Additionally measurement with multi channel audio need to take how we hear audio from different directional sources (i.e. surrounds) into account.
Loudness Management Strategies
§ “Measure and Set” § Measure content, set content with accurate metadata § “Measure and Scale” § Measure content, scale content to desired target level § “Target and Evaluate” § Select target loudness for submission – Specified in technical delivery specs § Evaluate content for compliance
Digital audio systems evolved from telecommunications technology developed in the 1930s. By the late 1960s, digital techniques offered benefits over analog for broadcast transmission. Digital audio works by sampling an analog audio signal at regular intervals, assigning it a binary code, and processing it as a digital data stream. Key aspects of digital audio include sampling rate, bit depth, anti-aliasing filters, pulse code modulation, quantization, multiplexing, dithering, bit rate, and digital clocking to ensure precise sampling.
Audio Essentials for Broadcast and MultiscreenEllis Reid
This wall chart highlights the key terms and standards required for the delivery of premium audio across broadcast and multiscreen workflows. It is designed as a quick reference for people who are responsible for delivering rich media experiences acress broadcast and over the top networks
This document discusses multimedia information representation and digitization principles. It covers the different media types used in multimedia like text, images, audio, and video. It explains how each media type is represented digitally and the encoding and decoding processes used to convert analog signals to digital and vice versa. It also discusses topics like digital sampling, quantization, signal bandwidth, encoding design, and image and text representation formats.
The document discusses the analog to digital conversion process. It explains that sounds are analog waves but computers are digital so an conversion is needed. The sound card contains an analog-to-digital converter (ADC) that samples sounds and converts them to binary digits and a digital-to-analog converter (DAC) that converts the digital signals back to analog waves for playback. The key parameters for conversion are the sampling rate, which must be over twice the highest frequency to avoid quality issues, and the bit depth, which determines the number of possible values and thus the resolution/quality. Higher rates and depths allow for better quality recordings.
This document discusses the process of sampling in signal processing. It defines key terms like analog and digital signals, sampling frequency, and samples. It explains how sampling works by taking regular measurements of a continuous signal's amplitude over time. This converts it into a discrete-time signal. It discusses applications of sampling like audio sampling, where signals are typically sampled above 20 kHz. It also discusses video sampling rates and speech sampling rates. The document contains examples and diagrams to illustrate these concepts.
The document provides an overview of an interactive voice conference on voice processing theory and algorithms for successful smart speakers and voice-enabled products. The agenda includes discussions on voice recognition algorithms, audio front-end processing, trigger word detection, beamforming, noise reduction, acoustic echo cancellation, and considerations for microphone and speaker integration in product design. Performance metrics and factors that affect various voice processing techniques are also outlined.
An audio quality evaluation of digital radio systemRojith Thomas
This document summarizes an evaluation of the audio quality of four digital radio systems: DAB, DAB+, HD Radio, and T-DMB. The evaluation used nine audio samples and tested each system at different bitrates. A group of listeners participated in two phases of subjective testing using the MUSHRA method. The results showed that certain bitrates for each system provided audio quality similar to or better than FM radio. Overall, the evaluation aimed to determine the optimal channel capacity needed to achieve a target audio quality for each digital radio system.
An audio quality evaluation of digital radio systemRojith Thomas
This document provides an audio quality evaluation of digital radio systems. It compares analog radio systems to proposed digital systems in terms of audio quality, channel capacity, spectrum efficiency and more. The evaluation involves collecting audio samples, encoding them using different codecs and bitrates, transmitting them over various digital radio systems, then having listeners rate the quality. The goal is to determine the optimal bitrate needed for a given audio quality across different digital radio systems.
An audio quality evaluation of digital radio systemRojith Thomas
The document evaluates the audio quality of four digital radio systems - DAB, DAB+, HD Radio, and T-DMB - at different bit rates using subjective testing. Nine audio samples were encoded and decoded at various bit rates for each system and tested using MUSHRA methodology. 21 listeners participated in the testing after passing pre-screening. Results showed mean audio quality for each system varied based on bit rate and audio content, with some systems performing better than others depending on the test conditions. The evaluation provides a reference for determining optimal bit rates to achieve target audio quality levels in digital radio systems.
An audio quality evaluation of digital radio systemRojith Thomas
1) The document evaluates the audio quality of 4 digital radio systems (DAB, DAB+, HD Radio, T-DMB) at different bitrates using MUSHRA subjective testing methods.
2) Phase 1 results show the mean audio quality as a function of bitrate for each system, identifying optimal bitrates to achieve a target quality.
3) Phase 2 confirms Phase 1 consistency by comparing similar quality groups across systems.
4) The evaluation provides reference results on the audio quality and channel capacity needed to achieve various quality targets for digital radio systems.
The document discusses analog and digital recording platforms. It states that analog and digital platforms are still standard in recording studios, with each having distinct sounds and applications in audio production. Analog recording uses magnetic tape that stores magnetic remnants representing audio signals. Digital recording represents audio as binary code by sampling amplitude over time at set bit rates. Both platforms remain important tools for music recording and production.
1. Analog-to-digital conversion (ADC) allows computers to interact with analog signals by sampling and quantizing analog signals from devices like CD players.
2. During recording, an ADC converts an analog audio signal into a digital format by repeatedly measuring and assigning a binary number to the signal's amplitude at set intervals defined by the sample rate.
3. During playback, a digital-to-analog converter (DAC) reconverts the digital numbers back into an analog signal by combining the amplitude information from each sample to rebuild the original wave.
The acoustic properties of a room design should ensure that it is easy to both speak and listen with a high degree of intelligibility. Reverberation Time is the single most important parameter used to evaluate room acoustics.
The document discusses digital audio and the process of digitizing sound. It explains that sound is converted to a stream of numbers through sampling and quantization. Sampling measures the amplitude of sound waves at regular time intervals, while quantization represents the measured amplitude with a finite number of digital values. For high quality audio, sampling rates of 44.1 kHz or higher and bit depths of 16 bits are commonly used. The document also covers topics like the Nyquist theorem, audio formats, editing digital audio, and more.
The document discusses the theory of audio, including:
1. What is audio and how it involves the production, recording, manipulation and reproduction of sound waves.
2. The basics of analog and digital audio, including how analog audio represents sound waves and how digital audio converts sound to binary numbers through sampling.
3. Key concepts in audio like bandwidth, which refers to the range of frequencies a signal occupies, and how analog audio is converted to digital audio through sampling and quantization.
This document provides 10 hints for making successful noise figure measurements. It discusses selecting an appropriate noise source, minimizing extraneous signals, reducing mismatch uncertainties, using averaging to minimize display jitter, avoiding non-linearities, accounting for mixer characteristics, using proper measurement corrections, choosing the optimal measurement bandwidth, accounting for path losses, and accounting for temperature variations of measurement components. The document is intended to help minimize uncertainties in noise figure measurements across a range of performance levels.
DDSP_2018_FOEHU - Lec 10 - Digital Signal Processing ApplicationsAmr E. Mohamed
The document provides an overview of digital signal processing applications including digital spectrum analysis, speech processing, and radar. It discusses topics like digital spectrum analyzers, speech production and perception, audio compression techniques including channel vocoding, ADPCM, LPC, and CELP coding. The key concepts covered are time-frequency analysis, the anatomy and acoustics of speech, speech and audio compression standards, and speech modeling and coding.
In this presentation, production of digital audio is discussed. Also brief introduction about digital audio broadcast, recording techniques and stereo phony is given.
2. The recognized
standard in audio test
Since 1985 AP has provided audio test
and measurement instruments to R&D
and production lines for every type of
audio device.
AP is the number one maker of audio
analyzers in the world.
Who is Audio Precision
3. Who uses AP?
" Audio Precision analyzers are used by makers
of all types of audio technology
! Pro audio
! A/V receivers
! Broadcast
! Smartphones
! Semiconductors
! Loudspeakers
! Microphones
! Automotive head units
! Blu-ray players
! MP3 players
! Bluetooth headsets
! Tablets and PCs
! Hearing aids
! Military
4. Monitoring vs. Measurement
4
Monitoring
" Displays key characteristics of actual program material in real time, such
as
! Level
! Channel activity
! Codec status
! Synchronization
Measurement
" Problem identification
! Noise, Distortion
! Frequency Response
! Glitches
" Proactive maintenance or used as needed
" Insures quality
…reveals very little of signal quality.
5. Measurement
5
" Studio maintenance
! Regular servicing
! Troubleshooting audio chain
! Evaluating new equipment
" On air broadcast quality of service
! Ensure broadcast quality across the service area outside the studio
6. Equipment maintenance
6
" Need the right interfaces & supported audio formats to evaluate
individual components
! AES/EBU interface
! High quality analog I/O
! SDI – can use an embedder / de-embedder if properly characterized
! Dolby E -- embed/deembed or “bit test”
9. In the studio
9
" Audio quality measurements
! Analog and digital
! Level
! THD+N
! Frequency response
! Phase
! Crosstalk
! Signal to noise ratio
! MOL
! Many more
" Signal chain and transport issues
! Encoding / decoding
! Digital and analog converters
! “Transparent” streaming of digital audio
! Bit truncation
! The transport itself (jitter)
16. Coded audio
16
" In television systems, multichannel streams may be coded in Dolby E
for studio transport, and Dolby Digital for emission.
" In such cases, the APx stimulus audio signal can be encoded and
passed down the chain of interest, and then can be measured after
decoding.
! Bit-accurate measurements will not provide useful measurements when passed
through an encode/decode cycle, which by design is not bit-accurate.
! Sine waves in coded audio
18. Radio Data System (RDS)
18
Narrow band signal with a 57 kHz sub-
carrier (3 x 19 kHz pilot tone)
30
Hz
15
kHz
23
kHz
38
kHz
53
kHz
Stereo Audio
(Left – Right)
19 kHz
Stereo
Pilot
(10%)
Mono Audio
(Left + Right)
58.7
kHz
RDS
(3%)
57
kHz
Our tech support engineers have developed a utility
to test FM RDS signals.
21. Mul=tone
" Single signal with multiple tones, typically 3,5,7 or 31 tones
" Extremely fast: 20+ results in <1 seconds
" Ideal for over the air broadcast and encoded formats
22. Mul=tone
22
The multitone is acquired and analyzed by FFT, providing many results.
The blue line represents a frequency response result aSer analysis.
The DUT has imposed its response on the mul=tone frequencies.
25. " FFTs (Fast Fourier Transform)
! The engine of modern audio analyzers
! Can derive most other measurements mathematically
" Look for
! 24-bit resolution
! Large max length 500K to 1M+ points
Today’s FFTs
24 bit APx monitor8-bit oscilloscope
27. Today’s FFTs
" Wide bandwidth option:
DC to 1MHz bandwidth @ 1.2M points = 2.38 Hz bin width
" Ideal for Class D amp design
! See switching frequencies
28. Mul=channel capability
" Multichannel analyzers can be
useful for maintenance in
broadcast and pro audio
environments.
" 16x faster than testing one
channel at a time
" One to many crosstalk
" See more channels on one screen
APx I/O varia2ons
• 2 x 2
• 2 x 4
• 2 x 8
• 8 x 8
• 8 x 16
30. UI, repor=ng and sharing data
" Analyzer should be easy to use
! Intuitive interface & real time monitors
! Easy automation with no coding
" Sharing data is as important as the research
! Share projects with limits and audio files with other departments anywhere
! Send results embedded in a project to compare or troubleshoot
! Database import / export
" Need fast, clear and easy reports
! Automatically generate PDF
! Customized Word documents
31. Calibra=on
" Similar, but very different
! Adjustment
! Calibration
! Accredited Calibration
" Traceability
! an unbroken chain of comparisons
! measurement uncertainty
! documentation
! competence
! reference to SI units
! calibration intervals
33. Bluetooth Case study: Hands-free Car Kit
" Engineer had a reference design for a Bluetooth car kit with HFP 1.6
(including the new wideband voice feature).
" His observation:
“When I’m connected to other Bluetooth devices in HFP, the audio I receive
from them sounds good.”
…
“But the audio they receive when I speak into the microphone sounds bad.”
34. Step 1: PESQ measurement
" Connect APx analog generator to his DUT and analyze speech signal
received over Bluetooth.
! PESQ: Perceptual Evaluation of Speech Quality
! Compares source and DUT for time alignment
! Uses speech samples
! Calculates MOS result
" Mean opinion score = 1.22
! Terrible!
! With wide-band voice,
MOS should be 4.0 to 4.5
MOS Quality Impairment
5 Excellent Impercep=ble
4 Good Percep=ble but not annoying
3 Fair Slightly annoying
2 Poor Annoying
1 Bad Very annoying
36. Step 3: Diagnos=c Tests
" Generate some pure sine signals through the DUT and observe the
results.
" In the following FFT spectrum plots, you can clearly see something
wrong.
40. Observa=ons from FFT spectra
" DUT has significant distortion due to a spurious tone at:
ftone = 8 kHz – fsignal
" Level of spurious tone increases as the frequency of the signal
" Note: 8 kHz = sample rate /2
Conclusion:
Clearly a DSP problem.
41. Bluetooth in – analog out?
" In Part 1, we looked at the signal transmitted by his DUT with analog
in and Bluetooth out.
" Next, we look at the opposite direction: Bluetooth in and analog out
(the signal from the phone calling the car kit).
Customer’s observation:
“In this direction, the audio sounds good.”
42. Step 4: PESQ measurement
" Connect APx Bluetooth generator to his DUT and analyze speech
signal received at analog input.
! Mean opinion score = 3.40
! Not bad, but should be better!
! With wide-band voice, MOS should be 4.0 to 4.5
45. Observa=on from Frequency Response
" Signal from DUT is being low-pass filtered at 4 kHz instead of 8 kHz
" This filter is designed for narrow band voice, not wide-band.
" Results in less clear sounding speech and a lower Mean Opinion
Score.
Conclusion:
Another DSP problem (wrong filter).
46. Conclusions …
" In ~ 1.5 hours, we were able to find and diagnose some serious
problems with this Bluetooth device.
" Using the “hello, …hello, … can you hear me now?” method, you
would never be able to find the problems or offer clues to fix.
47. For More Informa=on
47
" Free books:
! Measurement Techniques for Digital Audio
# ap.com/display/file/17
! The Audio Measurement Handbook
# ap.com/display/file/24
" Thomas Williams: tomw@ap.com
48. Why tes=ng is important
" How much is an hour of dead air?
" Do you want to be told about an error by a concerned listener?
Audio analyzer or sound card?
49. Audio analyzer or sound card?
" Trust & reliability
! Believe your own results
! Others believe your results
! Calibrated & traceable
Is the error you see
caused by your product
or your “analyzer”?
50. Audio analyzer or sound card?
" Productivity & sophisticated measurements
! No wasted time
! Built in measurements & proven algorithms
! Easy to share test data with others
! Professional technical support
Who do you call when
there’s a problem?
51. Common mistakes: grounding
" Most common mistake we see
" Star grounding is best
" Use heavy gauge and always check your grounds
The resistance in each leg of
the chain puts the devices at
different ground potentials, and
is not as effective as star
grounding.
52. Common mistakes: Class D filtering
" Class D switching amplifiers have unique problem of out of band noise
" Can cause inaccurate measurements, or even damage analyzer inputs
" Simple filter stops the problem
55. Common mistakes: Connec=ons
" Sounds simple, but it happens a lot
" Crossed cables
" Bad cables
" Analyzer set for wrong connections
" Always check your connections
" Use Loopback mode to confirm settings
56. 2700 Series
Most advanced analyzer for R&D
• Vanishingly low residual noise and THD
+N: —115 dB @ 2.0 Vrms
• True Dual Domain – analog and digital
• Generate and analyze a wide range of
waveforms
• API automation
• LabVIEW integration
• Chip-level connectivity with PSIA
• User-defined sweeps, switcher
support up to 192 channels
The world’s highest performance audio analyzer
THD+N of AP 2700 (Black) compared
with 5 other audio analyzers
57. APx Series
High performance for R&D
• Up to -110 dB THD+N
• Test Bluetooth, HDMI, I2S
• 1 million point FFT analyzer with
24 bit resolution from DC to 1MHz
• Multichannel and high bandwidth options
Multiple interfaces
• HDMI, Bluetooth, PDM
• AES/EBU, I2S, S/PDIF
Intuitive & easy to use
• One-click measurements
• Real-time monitors
• Automated reports
Fast, simple automation for production
• Up to 21 measurements in 1.2 seconds
• Automation without coding or use
the .NET API and LabVIEW driver
• Lockable projects with limits
Connectivity, Flexibility and Intuitive Operation