1) The document describes the transition in broadcasting from peak level normalization to loudness normalization of audio levels. This change addresses the problem of jarring jumps in audio levels between programs and commercials.
2) EBU Recommendation R 128 establishes a method for measuring and specifying loudness levels throughout the broadcast chain to help ensure consistent loudness. It introduces descriptors for program loudness, loudness range, and true peak level.
3) Loudness normalization aims to harmonize loudness levels within and between channels for listener benefit, while still allowing artistic variation within programs. It helps eliminate compromises previously needed to avoid sounding softer than louder competitors.
The Southeast Regional Acoustics Consortium (SEAC) met in March 2012 at Florida International University bringing together academic institutions, federal and regional fisheries and environmental management agencies, and private industry that conduct active acoustics research in the coastal environments of the US from North Carolina to Texas and the US Caribbean. Informal presentations and discussions highlighted the latest tools for fisheries research, organized around high-priority research objectives and management drivers (e.g., stock assessment improvements, integrated ecosystem assessments) and HTI’s Pat Nealson conducted a presentation to help demystify FM Slide/Chirp signals in hydroacoustics for fisheries assessments.
Impairment of signals & Transmission MediumAkila Dulaj
Nyquist Bit Rate Formula
Shannon Capacity
Wave Forms
Guided & Unguided Media
Difference Between UTP and STP
Different Modes of Fiber
Time Division Multiplexing
Frequency-Division Multiplexing
PSTN & POTS
ETC
Improvement of Phase Noise Compensation for Coherent OpticalOFDM via Data Ai...Raj Kumar Parihar
This document discusses using a data-aided phase equalizer (DAPE) to improve phase noise compensation in coherent optical orthogonal frequency-division multiplexing (OFDM) systems. DAPE utilizes both pilot symbols and receiver decisions to estimate channel factors and phase noise on a symbol-by-symbol basis. Simulation results show DAPE achieves a bit error rate of 10^-3 with 1-2.2 dB less required RF-pilot power compared to using RF pilots alone after 2000km fiber transmission at 40Gb/s. DAPE provides more accurate phase noise estimation, particularly for noisier scenarios, by averaging estimates across subcarriers.
1) Researchers developed a broadband erbium-doped fiber amplifier covering the C-band (1525-1565 nm) and L-band (1570-1610 nm) using a parallel structure of optimized C-band and L-band amplifiers.
2) The amplifier achieved an amplification bandwidth of over 70nm (1524-1602nm) with C-band average gain over 30dB and L-band gain variation less than 2dB.
3) By using separate C-band and L-band amplification and combining the signals, the design effectively prevented interaction between the two bands while simplifying the configuration.
The document discusses small-scale fading or multipath fading in wireless channels. It defines key terms like delay spread and Doppler spread caused by multipath propagation. It describes the time-varying channel impulse response and how multipath components combine constructively or destructively. It also introduces parameters used to characterize mobile multipath channels, such as coherence bandwidth, Doppler spread, and coherence time.
Single sideband modulation is more spectrally efficient than double sideband modulation by transmitting only one sideband. In single sideband modulation, there is a 90 degree phase offset between the sine and cosine signals. In practice, I/Q imbalance and carrier leakage are inevitable due to imperfections in the I/Q modulator and DC offsets. I/Q imbalance results in an undesired sideband and constellation distortion, degrading modulation accuracy. Carrier leakage arises from LO leakage and DC offsets, degrading SNR. Techniques like calibration circuits, balanced mixing, and adjusting I/Q gains/offsets can help minimize these impairments.
This document provides information about Doordarshan, the national public service broadcaster of India. It discusses the origin and development of Doordarshan from its beginnings in 1959 as a small studio to becoming one of the largest broadcast organizations in the world today. It also describes the technical processes and components involved in terrestrial television transmission using Doordarshan transmitters, including visual and audio signal chains, vestigial sideband transmission, and features of high power transmitters.
The PTP 230 is a point-to-point wireless Ethernet solution that offers high throughput and reliable broadband communications on a budget. It uses OFDM technology to provide robust performance even in non-line-of-sight conditions. The PTP 230 can synchronize multiple radios using GPS timing to prevent interference. It is available in the 5.4GHz and 5.8GHz bands and can triple throughput by replacing older PTP 100 radios.
The Southeast Regional Acoustics Consortium (SEAC) met in March 2012 at Florida International University bringing together academic institutions, federal and regional fisheries and environmental management agencies, and private industry that conduct active acoustics research in the coastal environments of the US from North Carolina to Texas and the US Caribbean. Informal presentations and discussions highlighted the latest tools for fisheries research, organized around high-priority research objectives and management drivers (e.g., stock assessment improvements, integrated ecosystem assessments) and HTI’s Pat Nealson conducted a presentation to help demystify FM Slide/Chirp signals in hydroacoustics for fisheries assessments.
Impairment of signals & Transmission MediumAkila Dulaj
Nyquist Bit Rate Formula
Shannon Capacity
Wave Forms
Guided & Unguided Media
Difference Between UTP and STP
Different Modes of Fiber
Time Division Multiplexing
Frequency-Division Multiplexing
PSTN & POTS
ETC
Improvement of Phase Noise Compensation for Coherent OpticalOFDM via Data Ai...Raj Kumar Parihar
This document discusses using a data-aided phase equalizer (DAPE) to improve phase noise compensation in coherent optical orthogonal frequency-division multiplexing (OFDM) systems. DAPE utilizes both pilot symbols and receiver decisions to estimate channel factors and phase noise on a symbol-by-symbol basis. Simulation results show DAPE achieves a bit error rate of 10^-3 with 1-2.2 dB less required RF-pilot power compared to using RF pilots alone after 2000km fiber transmission at 40Gb/s. DAPE provides more accurate phase noise estimation, particularly for noisier scenarios, by averaging estimates across subcarriers.
1) Researchers developed a broadband erbium-doped fiber amplifier covering the C-band (1525-1565 nm) and L-band (1570-1610 nm) using a parallel structure of optimized C-band and L-band amplifiers.
2) The amplifier achieved an amplification bandwidth of over 70nm (1524-1602nm) with C-band average gain over 30dB and L-band gain variation less than 2dB.
3) By using separate C-band and L-band amplification and combining the signals, the design effectively prevented interaction between the two bands while simplifying the configuration.
The document discusses small-scale fading or multipath fading in wireless channels. It defines key terms like delay spread and Doppler spread caused by multipath propagation. It describes the time-varying channel impulse response and how multipath components combine constructively or destructively. It also introduces parameters used to characterize mobile multipath channels, such as coherence bandwidth, Doppler spread, and coherence time.
Single sideband modulation is more spectrally efficient than double sideband modulation by transmitting only one sideband. In single sideband modulation, there is a 90 degree phase offset between the sine and cosine signals. In practice, I/Q imbalance and carrier leakage are inevitable due to imperfections in the I/Q modulator and DC offsets. I/Q imbalance results in an undesired sideband and constellation distortion, degrading modulation accuracy. Carrier leakage arises from LO leakage and DC offsets, degrading SNR. Techniques like calibration circuits, balanced mixing, and adjusting I/Q gains/offsets can help minimize these impairments.
This document provides information about Doordarshan, the national public service broadcaster of India. It discusses the origin and development of Doordarshan from its beginnings in 1959 as a small studio to becoming one of the largest broadcast organizations in the world today. It also describes the technical processes and components involved in terrestrial television transmission using Doordarshan transmitters, including visual and audio signal chains, vestigial sideband transmission, and features of high power transmitters.
The PTP 230 is a point-to-point wireless Ethernet solution that offers high throughput and reliable broadband communications on a budget. It uses OFDM technology to provide robust performance even in non-line-of-sight conditions. The PTP 230 can synchronize multiple radios using GPS timing to prevent interference. It is available in the 5.4GHz and 5.8GHz bands and can triple throughput by replacing older PTP 100 radios.
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.
This document provides an overview of digital communications and source encoding. It discusses why digital communication is preferable to analog, describes the basic block diagram of a digital communication system, and defines key terms like the information source and channel encoder. The document then covers some of the foundational work in digital communications, including Nyquist's sampling theorem and Shannon's channel capacity theorem. It discusses ideal sampling and the Nyquist rate, as well as practical sampling techniques like sample-and-hold. The document also covers quantization, quantization noise, and how the step size and number of quantization levels affect the signal-to-quantization noise ratio. In closing, it briefly mentions pulse code modulation and nonuniform quantization.
This document summarizes an experiment on a radio-over-fiber system for hybrid wireless-optical broadband access networks. The system used 5 GHz radio frequency signals transmitted over a 60 km fiber link using 12.5 GHz dense wavelength division multiplexing with 5 channels. Digital coherent detection and Raman amplification were used to provide enhanced power budget and reach. Bit error rate measurements showed error-free transmission was achieved after the 60 km link with and without neighboring channel interference, demonstrating the feasibility of the system for long-reach hybrid wireless-optical networks.
This document proposes an OFDM receiver capable of estimating and correcting the subcarrier dependent Doppler shifting on a symbol-by-symbol basis for highly mobile underwater acoustic networks. It presents a receiver structure that estimates the Doppler rate for each OFDM symbol based on the recovered data from the previous symbol, allowing for compensation of changing Doppler rates. Two algorithms are developed for the symbol-by-symbol Doppler rate estimation: one ad-hoc and one based on the marginal maximum likelihood principle. Simulation results show the receiver design performs well for a source accelerating at 0.29 m/s^2.
Design and implementation of test bench for frequency modulation and demodula...Karrar Abd Alhadi
This document presents a project on designing and implementing a test bench for frequency modulation and demodulation. It contains an introduction to FM radio technology and its history. It discusses topics such as modulation index, Bessel functions, FM power distribution, and average power. It also covers the principles and methods of FM modulation and demodulation, including narrowband FM, wideband FM modulators, and FM demodulator classifications. The project aims to build circuit diagrams for an FM modulator and demodulator. It analyzes the results of experiments on frequency modulation and demodulation. The document contains acknowledgments, an abstract, table of contents, and references.
The document provides an overview of ATSC Mobile TV and how it works. It discusses the challenges of mitigating mobile fading and maintaining backward compatibility with legacy receivers. It introduces the dual stream system used by ATSC Mobile TV to transmit an HDTV/SDTV stream and a separate mobile stream. It describes the cross-layer design of ATSC Mobile TV, which uses new channel coding, training signals, and signaling to address mobile fading while maintaining compatibility. It also outlines the basic blocks of the ATSC Mobile TV transmitter and receiver.
The document discusses the process of mastering music, including defining mastering as tweaking the final stereo mix. It covers topics like the loudness war where increased compression is used to make recordings louder, new standards for measuring loudness like LUFS, considerations for dynamic range and headroom, tools used in mastering like compression and EQ, and the importance of monitoring on different systems when finalizing a track.
1) The Fagor MicroMATV pro 100 is a programmable selective amplifier that allows combining signals from up to six different antenna inputs and distributing the signal to TV sets or a cable network.
2) The amplifier provides inputs for FM, VHF bands I/III, and three UHF inputs, and allows setting filters and gain for each input band.
3) In testing, the amplifier showed nearly perfect frequency responses for the FM and VHF bands, though some gain and filter specifications were exceeded. UHF inputs also demonstrated good flat frequency responses.
1. The document provides definitions and explanations of key concepts in digital communication systems and coding techniques. It contains questions and answers on topics like modulation, sampling theorem, equalization, coding, and error correction.
2. Key concepts defined include modulation, inter-symbol interference, matched filters, maximum likelihood detection, orthogonal signals, coding gain, and convolutional codes.
3. Error correction techniques like Hamming codes, BCH codes, and RS codes are described in terms of their parameters, capabilities, and decoding methods. The differences between block codes and convolutional codes are also summarized.
The document tests and evaluates LNBs from GT-SAT International's Diamond Line series. Measurement results show the Diamond Line LNBs provide significantly stronger output power than a reference LNB, compensating for signal attenuation over 30-50 meters of coax cable. Noise performance tests find the Diamond Line models are equal to or better than the reference LNB. The increased output power is achieved without degradation of noise performance. Current consumption is slightly higher for the high-power Diamond Line LNBs but expected given their capabilities. Overall, the Diamond Line LNBs perform well in output power and noise tests, making them suitable for difficult signal reception areas.
DAB (Digital Audio Broadcasting) is a digital broadcasting system for radio that provides listeners with more choice, better sound quality, and more information than analog radio. It allows broadcasters to transmit many more stations using the same frequency, and listeners can receive between 30-50 stations in major cities. DAB delivers improved sound quality by locking onto the strongest signal and eliminating hiss, crackle, and fade. Digital radios also allow listeners to easily tune to stations by name, display text information from stations, and pause, rewind, and record live radio.
Effect of Pump Dithering at Each Stage of Cascaded Fiber Optical Parametric A...TELKOMNIKA JOURNAL
Cascaded fiber optical parametric amplifier (FOPA) can enhance gain and bandwidth. The gain and bandwidth can be further enhanced by dithering the FOPA pump. However, to our knowledge, the effects of a pump dithering at every stage of cascaded FOPA have not been discussed. The study of performance at every stage of cascaded FOPA is quite interesting and beneficial in designing the system. Here, we analyzed, using OptiSystem software, each stage of a cascaded FOPA, when there was a pump dithering and not. The results showed that the pump dithering enhanced the gain and broaden the bandwidth at every stage. The gain and bandwidth obtained with the pump dithering were 27 dB and 20 nm, respectively. On the other hand, when there was no pump dithering, the gain and bandwidth were 9 dB and 12 nm, respectively.
This document discusses frequency modulation (FM) and provides details about:
1) FM can be used for both analog and digital data transmission by varying the instantaneous frequency of a carrier wave.
2) In analog FM the carrier frequency varies continuously, while in digital FM it shifts abruptly between discrete frequency states.
3) FM bandwidth depends on the modulation index, with higher indices resulting in wider bandwidth signals classified as wideband FM.
This white paper compares two technologies for optimizing satellite capacity efficiency for IP trunking networks: adaptive coding and modulation (ACM) and signal cancellation. It analyzes a reference network delivering symmetrical and asymmetrical services from a hub in Sudan to 10 remote sites in Africa. Using conventional fixed modulation, a minimum of 3 transponders is needed for symmetrical services but asymmetrical services require too much bandwidth to fit in two transponders. The paper will evaluate if ACM or signal cancellation can provide bandwidth savings.
Newtec - Optimisation of Satellite Capacity Efficiency for IP Trunking Applic...Sematron UK Ltd
This white paper compares two technologies for optimizing satellite capacity efficiency for IP trunking networks: adaptive coding and modulation (ACM) and signal cancellation. It analyzes a reference network delivering symmetrical and asymmetrical services from a hub in Sudan to 10 remote sites in Africa. Using conventional fixed modulation, a minimum of 3 transponders is needed for symmetrical services but asymmetrical services require too much bandwidth to fit in two transponders. The paper will evaluate if ACM or signal cancellation can provide a more efficient implementation.
This document provides an overview of angle modulation techniques, including phase modulation and frequency modulation. It defines these terms and discusses how they work. It also covers topics like FM bandwidth, generation of narrowband and wideband FM signals, FM detection methods, and noise analysis for AM and FM signals. Key points include: Angle modulation varies the angle of the carrier wave according to the baseband signal while keeping amplitude constant. It can provide better noise performance than AM at the cost of increased bandwidth. Narrowband FM has a bandwidth of about 2fm while wideband FM bandwidth is about 2Δf. FM detection can be done with differentiation and envelope detection or using a phase locked loop with a VCO.
This document provides an overview of communication basics and amplitude modulation. It discusses how communication involves transmitting and receiving information, and how modulation translates signals to higher frequencies for long-distance transmission. It then describes various amplitude modulation techniques like AM, DSB, and SSB. Key aspects covered include the AM envelope, frequency spectrum of AM waves, AM modulation indexes, and different AM modulation and demodulation methods.
This document discusses different types of video signals and coding standards. It describes component video which uses separate signals for red, green, and blue channels, providing the best color reproduction. Composite video mixes color and intensity into a single signal, causing some interference. S-Video uses two signals for luminance and composite chrominance, reducing crosstalk. Digital video allows storage, access and editing of video and is more tolerant of noise. Standards like CCIR 601 set component digital video parameters. High definition TV aims to increase visual field width through greater pixel counts and aspect ratios. The ATSC digital TV standard supports various formats up to 1080p at 60 frames/sec. MPEG-2 is used for video compression and AC-3
This project aims to design and develop a multi-format video converter to convert analog video signals from target detection systems to digital formats for transmission to and processing by computing devices. The converter will convert analog video signals to compressed digital formats and transfer the data over USB or Ethernet ports. The development of this converter is motivated by the benefits of digital signal transmission, storage, and processing compared to analog.
AKOCONTROL - DARWIN is a new generation of controllers for industrial and commercial refrigeration introduced by AKO. The standard DARWIN controller is suitable for commercial refrigeration applications with up to 4 relays. It offers features like ultra fast setup, application-specific browsing, and signal icons for diagnosis. Customized DARWIN controllers can be seamlessly integrated into refrigeration cabinets with comprehensive solutions and energy savings up to 23%. AKO aims to evolve with market demands to offer high quality products meeting customer needs.
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.
This document provides an overview of digital communications and source encoding. It discusses why digital communication is preferable to analog, describes the basic block diagram of a digital communication system, and defines key terms like the information source and channel encoder. The document then covers some of the foundational work in digital communications, including Nyquist's sampling theorem and Shannon's channel capacity theorem. It discusses ideal sampling and the Nyquist rate, as well as practical sampling techniques like sample-and-hold. The document also covers quantization, quantization noise, and how the step size and number of quantization levels affect the signal-to-quantization noise ratio. In closing, it briefly mentions pulse code modulation and nonuniform quantization.
This document summarizes an experiment on a radio-over-fiber system for hybrid wireless-optical broadband access networks. The system used 5 GHz radio frequency signals transmitted over a 60 km fiber link using 12.5 GHz dense wavelength division multiplexing with 5 channels. Digital coherent detection and Raman amplification were used to provide enhanced power budget and reach. Bit error rate measurements showed error-free transmission was achieved after the 60 km link with and without neighboring channel interference, demonstrating the feasibility of the system for long-reach hybrid wireless-optical networks.
This document proposes an OFDM receiver capable of estimating and correcting the subcarrier dependent Doppler shifting on a symbol-by-symbol basis for highly mobile underwater acoustic networks. It presents a receiver structure that estimates the Doppler rate for each OFDM symbol based on the recovered data from the previous symbol, allowing for compensation of changing Doppler rates. Two algorithms are developed for the symbol-by-symbol Doppler rate estimation: one ad-hoc and one based on the marginal maximum likelihood principle. Simulation results show the receiver design performs well for a source accelerating at 0.29 m/s^2.
Design and implementation of test bench for frequency modulation and demodula...Karrar Abd Alhadi
This document presents a project on designing and implementing a test bench for frequency modulation and demodulation. It contains an introduction to FM radio technology and its history. It discusses topics such as modulation index, Bessel functions, FM power distribution, and average power. It also covers the principles and methods of FM modulation and demodulation, including narrowband FM, wideband FM modulators, and FM demodulator classifications. The project aims to build circuit diagrams for an FM modulator and demodulator. It analyzes the results of experiments on frequency modulation and demodulation. The document contains acknowledgments, an abstract, table of contents, and references.
The document provides an overview of ATSC Mobile TV and how it works. It discusses the challenges of mitigating mobile fading and maintaining backward compatibility with legacy receivers. It introduces the dual stream system used by ATSC Mobile TV to transmit an HDTV/SDTV stream and a separate mobile stream. It describes the cross-layer design of ATSC Mobile TV, which uses new channel coding, training signals, and signaling to address mobile fading while maintaining compatibility. It also outlines the basic blocks of the ATSC Mobile TV transmitter and receiver.
The document discusses the process of mastering music, including defining mastering as tweaking the final stereo mix. It covers topics like the loudness war where increased compression is used to make recordings louder, new standards for measuring loudness like LUFS, considerations for dynamic range and headroom, tools used in mastering like compression and EQ, and the importance of monitoring on different systems when finalizing a track.
1) The Fagor MicroMATV pro 100 is a programmable selective amplifier that allows combining signals from up to six different antenna inputs and distributing the signal to TV sets or a cable network.
2) The amplifier provides inputs for FM, VHF bands I/III, and three UHF inputs, and allows setting filters and gain for each input band.
3) In testing, the amplifier showed nearly perfect frequency responses for the FM and VHF bands, though some gain and filter specifications were exceeded. UHF inputs also demonstrated good flat frequency responses.
1. The document provides definitions and explanations of key concepts in digital communication systems and coding techniques. It contains questions and answers on topics like modulation, sampling theorem, equalization, coding, and error correction.
2. Key concepts defined include modulation, inter-symbol interference, matched filters, maximum likelihood detection, orthogonal signals, coding gain, and convolutional codes.
3. Error correction techniques like Hamming codes, BCH codes, and RS codes are described in terms of their parameters, capabilities, and decoding methods. The differences between block codes and convolutional codes are also summarized.
The document tests and evaluates LNBs from GT-SAT International's Diamond Line series. Measurement results show the Diamond Line LNBs provide significantly stronger output power than a reference LNB, compensating for signal attenuation over 30-50 meters of coax cable. Noise performance tests find the Diamond Line models are equal to or better than the reference LNB. The increased output power is achieved without degradation of noise performance. Current consumption is slightly higher for the high-power Diamond Line LNBs but expected given their capabilities. Overall, the Diamond Line LNBs perform well in output power and noise tests, making them suitable for difficult signal reception areas.
DAB (Digital Audio Broadcasting) is a digital broadcasting system for radio that provides listeners with more choice, better sound quality, and more information than analog radio. It allows broadcasters to transmit many more stations using the same frequency, and listeners can receive between 30-50 stations in major cities. DAB delivers improved sound quality by locking onto the strongest signal and eliminating hiss, crackle, and fade. Digital radios also allow listeners to easily tune to stations by name, display text information from stations, and pause, rewind, and record live radio.
Effect of Pump Dithering at Each Stage of Cascaded Fiber Optical Parametric A...TELKOMNIKA JOURNAL
Cascaded fiber optical parametric amplifier (FOPA) can enhance gain and bandwidth. The gain and bandwidth can be further enhanced by dithering the FOPA pump. However, to our knowledge, the effects of a pump dithering at every stage of cascaded FOPA have not been discussed. The study of performance at every stage of cascaded FOPA is quite interesting and beneficial in designing the system. Here, we analyzed, using OptiSystem software, each stage of a cascaded FOPA, when there was a pump dithering and not. The results showed that the pump dithering enhanced the gain and broaden the bandwidth at every stage. The gain and bandwidth obtained with the pump dithering were 27 dB and 20 nm, respectively. On the other hand, when there was no pump dithering, the gain and bandwidth were 9 dB and 12 nm, respectively.
This document discusses frequency modulation (FM) and provides details about:
1) FM can be used for both analog and digital data transmission by varying the instantaneous frequency of a carrier wave.
2) In analog FM the carrier frequency varies continuously, while in digital FM it shifts abruptly between discrete frequency states.
3) FM bandwidth depends on the modulation index, with higher indices resulting in wider bandwidth signals classified as wideband FM.
This white paper compares two technologies for optimizing satellite capacity efficiency for IP trunking networks: adaptive coding and modulation (ACM) and signal cancellation. It analyzes a reference network delivering symmetrical and asymmetrical services from a hub in Sudan to 10 remote sites in Africa. Using conventional fixed modulation, a minimum of 3 transponders is needed for symmetrical services but asymmetrical services require too much bandwidth to fit in two transponders. The paper will evaluate if ACM or signal cancellation can provide bandwidth savings.
Newtec - Optimisation of Satellite Capacity Efficiency for IP Trunking Applic...Sematron UK Ltd
This white paper compares two technologies for optimizing satellite capacity efficiency for IP trunking networks: adaptive coding and modulation (ACM) and signal cancellation. It analyzes a reference network delivering symmetrical and asymmetrical services from a hub in Sudan to 10 remote sites in Africa. Using conventional fixed modulation, a minimum of 3 transponders is needed for symmetrical services but asymmetrical services require too much bandwidth to fit in two transponders. The paper will evaluate if ACM or signal cancellation can provide a more efficient implementation.
This document provides an overview of angle modulation techniques, including phase modulation and frequency modulation. It defines these terms and discusses how they work. It also covers topics like FM bandwidth, generation of narrowband and wideband FM signals, FM detection methods, and noise analysis for AM and FM signals. Key points include: Angle modulation varies the angle of the carrier wave according to the baseband signal while keeping amplitude constant. It can provide better noise performance than AM at the cost of increased bandwidth. Narrowband FM has a bandwidth of about 2fm while wideband FM bandwidth is about 2Δf. FM detection can be done with differentiation and envelope detection or using a phase locked loop with a VCO.
This document provides an overview of communication basics and amplitude modulation. It discusses how communication involves transmitting and receiving information, and how modulation translates signals to higher frequencies for long-distance transmission. It then describes various amplitude modulation techniques like AM, DSB, and SSB. Key aspects covered include the AM envelope, frequency spectrum of AM waves, AM modulation indexes, and different AM modulation and demodulation methods.
This document discusses different types of video signals and coding standards. It describes component video which uses separate signals for red, green, and blue channels, providing the best color reproduction. Composite video mixes color and intensity into a single signal, causing some interference. S-Video uses two signals for luminance and composite chrominance, reducing crosstalk. Digital video allows storage, access and editing of video and is more tolerant of noise. Standards like CCIR 601 set component digital video parameters. High definition TV aims to increase visual field width through greater pixel counts and aspect ratios. The ATSC digital TV standard supports various formats up to 1080p at 60 frames/sec. MPEG-2 is used for video compression and AC-3
This project aims to design and develop a multi-format video converter to convert analog video signals from target detection systems to digital formats for transmission to and processing by computing devices. The converter will convert analog video signals to compressed digital formats and transfer the data over USB or Ethernet ports. The development of this converter is motivated by the benefits of digital signal transmission, storage, and processing compared to analog.
AKOCONTROL - DARWIN is a new generation of controllers for industrial and commercial refrigeration introduced by AKO. The standard DARWIN controller is suitable for commercial refrigeration applications with up to 4 relays. It offers features like ultra fast setup, application-specific browsing, and signal icons for diagnosis. Customized DARWIN controllers can be seamlessly integrated into refrigeration cabinets with comprehensive solutions and energy savings up to 23%. AKO aims to evolve with market demands to offer high quality products meeting customer needs.
Digital storytelling can benefit students with disabilities in several ways. It allows for accommodations and adaptations that increase accessibility of the curriculum. Students can both access pre-made digital stories that teach social skills, as well as create their own stories. This fosters language arts skills like editing and encourages creative expression. Studies found students on the autism spectrum displayed similar narrative skills to peers when using digital tools. Digital storytelling motivates students and increases confidence by allowing them to learn and express themselves in a comfortable way.
Le rapport 2013 du WEF sur les risques dans le mondeLeSoir.be
This document is the introduction to the World Economic Forum's 2013 report on global risks. It provides information on the sources of the report's data and analysis, but notes that the information has not been independently verified. The introduction also states that the companies contributing to the report operate in a changing environment and disclaim any liability for losses arising from the use of the report's information. It lists the editor-in-chief and contributing organizations to the 2013 Global Risks report.
The document discusses the history and development of loudness measurement standards for digital television and audio. It describes how digital audio's greater dynamic range led to issues with commercials being perceived as louder than programs. This prompted the development of loudness measurement methods like LKFS (Loudness, K-Weighting, referenced to digital full scale) by organizations like ITU, EBU, ATSC, and others. Governments and regulators worldwide have since implemented loudness standards and legislation to address viewer complaints about loud commercials. Modern broadcast monitoring systems help ensure compliance with these standards.
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
The document describes Sage Instruments' Echo Sounder and Echo Generator tools. The Echo Sounder measures multiple echoes in terms of level and delay to characterize voice quality. It uses a code-domain reflectometry technique to detect echoes with levels below -60 dB. The Echo Generator generates programmable echoes to test echo cancellers and measure network delays and losses. Together these tools provide a way to test and measure important voice quality factors like echoes in telephone networks.
A sound level meter is an instrument used to objectively measure sound pressure levels in decibels in a standardized way, similar to how the human ear perceives sound. It has a microphone that converts sound to electrical signals, and uses time and frequency weightings to process the signals in a way that approximates human hearing. The processed signal is then amplified and its root mean square value determined to indicate the sound level. Sound level meters are used to measure noise from sources like industry, traffic, and construction to evaluate noise environments.
HUFFMAN CODING ALGORITHM BASED ADAPTIVE NOISE CANCELLATIONIRJET Journal
This document presents a paper that proposes using Huffman coding and adaptive noise cancellation algorithms together to reduce noise while transmitting audio and visual signals. It begins with an abstract that discusses using data compression and algorithms to decrease the impact of background noise on recorded signals while maintaining the original undisturbed form of the signals. It then provides background on human hearing capabilities and digital audio signals. The document discusses existing noise cancellation systems and their limitations. It proposes a new framework that uses Huffman coding to create an adaptive code for each unique sound component, and builds a Huffman tree from those codes to map codes to probability. The proposed system is claimed to better remove signal transients and remaining noise artifacts compared to existing short-time Fourier transform approaches.
20150211 NAB paper - Audio Loudness Range -John KeanJeremy Adams
This study investigated the range of audio loudness preferred by listeners in different ambient noise environments. Listeners adjusted playback volume levels of music, news, and talk samples played back in simulated noise conditions like homes, offices, cars and public transit. Results could inform automatic loudness control systems to better match listeners' preferences across environments. Testing involved 37 participants adjusting volume levels of samples played through speakers and earbuds in a simulated noise laboratory.
This document summarizes the rationale for Localization Cue Correction (LCC) technology, which aims to address issues with traditional stereo playback using widely spaced speakers. It discusses how LCC positions speakers closer together to provide more natural localization cues, thereby creating a more immersive soundstage. The technology is presented as overcoming limitations of other virtual surround sound systems by accurately reproducing interaural level and time differences without additional processing. Benefits highlighted include improved off-axis listening, reduced need for room calibration, and advantages for content creators in mixing and mastering.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
Satellite communications by dennis roddy (4th edition)Adam Năm
This document discusses frequency modulation (FM) bandwidth calculations and signal-to-noise ratios. It contains examples of calculating bandwidth for different FM signals using Carson's rule and other equations. It also examines how FM detection provides "processing gain" which improves the signal-to-noise ratio by spreading the noise power over a wider bandwidth, while the signal power remains concentrated. Threshold effects and carrier-to-noise ratios are also discussed.
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1. LOUDNESS
On the way to
Loudness nirvana
— audio levelling with EBU R 128
Florian Camerer
ORF
This article describes one of the most
fundamental changes in the history of audio Loudness normalisation
in broadcasting: the change of the levelling is a true audio-levelling
paradigm from peak normalisation to loudness
normalisation. This change is vital because
revolution!
of a problem that has become a major
source of irritation for television and radio audiences around the world – that of the
jump in audio levels at the breaks within programmes, between programmes and
between channels. Loudness normalisation is the solution to counteract this
problem.
EBU Recommendation R 128 [1] establishes a predictable and well-defined method to measure
the loudness1 level for news, sports, advertisements, drama, music, promotions, films etc., through-
out the broadcast chain, and thereby helps professionals to create robust specifications for ingest,
production, playout and distribution to a multitude of platforms. Four supporting documents have
been created by the EBU to aid the audio industry to
work with Rec. R 128. It is based entirely on open
standards and aims to harmonise the way we produce
and measure audio internationally. In addition to Pro-
gramme Loudness, R 128 introduces two more audio
descriptors, Loudness Range and Maximum True
Peak Level. All three are designed to work together,
forming a set of essential descriptors that characterise
an audio signal.
Loudness metering and loudness normalisation sig-
nify a true audio levelling revolution (see Fig. 1). Fur-
Figure 1
thermore, this loudness-levelling paradigm affects all
Peak-level normalisation vs. Loudness-level
normalisation of a series of programmes stages of an audio broadcast signal, from acquisition to
distribution and transmission. Thus, the ultimate goal is
to harmonise audio loudness levels within broadcast
channels as well as between channels to achieve an equal universal loudness level for the benefit of
the listeners. To be clear: the loudness level can (and should!) still vary according to artistic and tech-
1. Loudness refers to the perceived strength of a piece of audio (music, speech, sound effects etc.). The
loudness depends on the level, frequency, content and the duration of the audio, amongst other things.
EBU TECHNICAL REVIEW – 2010 Q3 1/9
F. Camerer
2. LOUDNESS
nical needs within a programme. Rather, the loudness normalisation method uses the average loud-
ness of a programme, to make the level consistent between programmes and between channels.
The experience of several EBU Members has shown that working with the loudness paradigm is
both liberating and satisfactory. The fight for “Who is the loudest” disappears, mixes can be more
dynamic, there are fewer dynamic compression artefacts, such as “pumping”, and thus there is an
overall increase of audio quality! Programme makers who favoured dynamic mixes in the past
are now relieved from potential compromises because their programme no longer sounds softer
than more-compressed ones. With loudness normalisation, this compromise is gone. Nirvana is
closer than ever!
The origin of the problem – the “loudness war”
Audio metering in broadcasting today is based typically on quasi-peak programme meters (QPPMs
– more usually known as just PPMs). It is “quasi” because of its finite reaction time of 10 ms
(although 5 ms is also found). In practice, this
0 means that signal peaks shorter than this reaction
Level [dBFS ] Dev [kHz] headroom
time won’t be displayed correctly, if at all (for
0 -4.6
loss example, transients such as those created when
keys are jangled). In order to provide headroom
-4.6 50 -9 50
lo u d n e ss for these transients, which one wouldn’t see on
-9 ga in the meter, but which should nevertheless be there
30 30
so as to contribute to the “openness” of the audio
-20 signal, the agreed Permitted Maximum Level
(PML) was set at –9 dBFS.
-20
This value was based on the familiar – and in
-30 many places, still extant – method of delivering
sound to the home on an FM carrier. The carrier's
-30
maximum deviation for TV was standardized in
many countries at 50 kHz and the PML at 30 kHz
deviation (equating to –9 dBFS), which thus
allowed 20 kHz, or 4.4 dB of headroom.
Figure 2
Relationship between Permitted Maximum Lev- However, commercial pressures have taken a
hold and the response to the pressure to stand
el (with QPPM) and FM Deviation and the abuse
of it: gaining loudness, but losing headroomout from the competition has been to be louder
than it. Modern peak metering and powerful
dynamic-range processing allowed organizations
to realign the PML to equate to the maximum deviation (50 kHz) of the FM carrier (see Fig. 2). All
the transients have to be chopped at the PML to avoid distortion, but that has been thought an
acceptable compromise by those who have implemented it.
When someone at home switches between one of these stations and a station which has not joined
the “war” or is taken to a loud advertisement in a well-balanced programme with a wider dynamic
range, the audio level jumps and there is a grab for the remote control to adjust the sound to a more
acceptable level. In the case of loud advertisements, the sound then has to be re-adjusted when the
main programme returns. It is no wonder that so many complaints are received by the broadcasters.
Other people solve the problem by muting the sound during the advertisements and so the message
is at least greatly diluted.
A Standard Emerges – and the EBU develops it
The International Telecommunication Union (ITU) recognized the problem and its work gave rise to
ITU-R BS.1770 [2]. The purpose of that standard was to establish an agreed algorithm for the
measurement of loudness and the true peak levels of programmes. It is a robust standard which
has the benefit of a simple implementation. In brief, it defines a “K-weighting” curve (a modified sec-
EBU TECHNICAL REVIEW – 2010 Q3 2/9
F. Camerer
3. LOUDNESS
ond-order high-pass filter)
which forms the basis for
matching an inherently subjec-
tive impression with an objec-
tive measurement.
This curve is applied to all the
channels (except the Low-Fre-
quency Effects (LFE) channel
which is discarded from the
measurement), the total mean
square energy is then calcu-
lated (with different gain fac-
tors for the front and surround
channels, see Fig. 3) and the
result is displayed as “LKFS”
(Loudness, K-Weighting, refer-
enced to digital Full Scale).
For relative measurements,
Loudness Units (LU) are used,
where 1 LU is equivalent to
1 dB.
A more detailed study of the
algorithm can be found in ITU-
R BS.1770 [2] as well as in
EBU Tech Doc. 3343 – “Practi-
cal Guidelines” [3]).
BS.1770 also defines and rec-
ommends the use of a true Figure 3
Channel processing and summation in ITU-R BS.1770
peak meter for measuring
peaks. Such a meter runs at a
higher sampling rate than the audio signal (usually 4x oversampling) to catch inter-sample peaks
which might otherwise exceed 0 dBFS and thus cause distortion later in the chain.
ITU-R BS.1770 provides the basis for
R 128 and ITU-R BS.1770 are the EBU Recommendation R 128 which
basis. Four more EBU technical extends the ITU standard by actually
defining a specific Target Level (see
documents provide details. below) for loudness normalisation as
well as a gating method to improve the
loudness matching of programmes which contain longer periods of silence or isolated utterances.
The EBU’s development was required to accommodate the needs of programme makers, with par-
ticular regard to having a means to measure complete mixes (rather than just one component, such
as the dialogue) and the loudness range of the programme. To do this, the EBU has specified three
new parameters:
Programme Loudness
Loudness Range
True Peak Level
Programme Loudness
Programme Loudness describes the long-term integrated loudness over the duration of a pro-
gramme. (In R 128, the definition of the word “programme” is also used to refer to advertisements,
EBU TECHNICAL REVIEW – 2010 Q3 3/9
F. Camerer
4. LOUDNESS
bumpers and other interstitials.) The parameter consists of one number (in LUFS2), with one
number after the decimal point) which indicates “how loud the programme is on average”. This is
measured with a meter compliant to ITU-R BS.1770 with the addition of a gating function. The gate
serves to pause the loudness measurement when the signal drops below a certain threshold. With-
out this gating function, programmes with longer periods of silence or low-level background sounds
or noise will get too low a value for the integrated loudness level. Such programmes would subse-
quently be too loud when broadcast.
Following a series of tests, a gate of –8 dB relative to the ungated LUFS measurement, with a block
length of 400 ms, was agreed. The tests also confirmed, along with other findings, the choice of the
Target Level to which every audio signal will be normalised. It is:
–23 LUFS (–8 rel gate)
A deviation of ± 1 LU is acceptable for pro-
–23 LUFS is the new centre of grammes where an exact normalisation to
the audio-levelling universe. the Target Level of –23 LUFS is not
achievable practically (such as live pro-
grammes or ones which have an exceed-
ingly short turn-round). In cases where the levels of a programme’s individual signals are to a large
extent unpredictable, or where a programme consists of only background elements (for example,
the music bed for a weather programme), this tolerance may be too tight. It is therefore anticipated
for such cases that the integrated Loudness Level may lie outside the tolerance specified by R 128.
Loudness Range
Another major consideration was the Loudness Range which would be needed to accommodate all
programmes (provided that they don’t exceed the tolerable loudness range for domestic listening).
The Loudness Range (LRA) descriptor quantifies (in LU) the variation of the loudness measure-
ment of a programme. It is based on
the statistical distribution of loudness Loudness Range is a generic
within a programme, thereby exclud-
ing the extremes. Thus, for example, descriptor that helps to decide if
a single gunshot is not able to bias dynamic compression is necessary.
the outcome of the LRA computation.
EBU Recommendation R 128 does not specify a maximum permitted LRA, as it is dependent on fac-
tors such as the tolerance window of the average listener to the station, the distribution of genres of
the station etc. R 128 does, however, strongly encourage the use of LRA to determine if dynamic
treatment of an audio signal is needed and to match the signal with the requirements of a particular
transmission channel or platform. More details about LRA may be found in EBU Tech Doc 3342 [4].
First experiences at broadcasting stations suggest a maximum LRA value of approximately 20 LU
for highly-dynamic material, such as action movies or classical music. The majority of programming
will never need to fully use such a high LRA value or, indeed, be able to reach it!
True Peak Level
The True Peak Level of an audio signal indicates the maximum (positive or negative) value of the
signal waveform in the continuous time domain; this value is, in most cases, higher than that shown
by a quasi-peak meter or even a sample-peak meter, both of which would miss the true peaks which
potentially lie between samples. The use of an oversampling meter, which is compliant to BS.1770,
allows those peaks to be detected.
2. LUFS indicates the value of K-weighted loudness with reference to digital full scale. The EBU recom-
mends this unit to overcome an inconsistency between ITU-R BS.1770 and ITU-R BS.1771. This unit
complies with ISO 80000-8.
EBU TECHNICAL REVIEW – 2010 Q3 4/9
F. Camerer
5. LOUDNESS
An oversampling meter may still slightly
under-read the actual peak value Oversampling peak meters
(depending on the oversampling fre- provide a good estimate of the
quency) and so the Maximum Permit-
ted True Peak Level for Production is: true peak of an audio signal.
–1 dBTP
Sample peak meters don’t.
Note that some parts of the chain, such as analogue re-broadcasters and users of low-bitrate coders
require a lower True Peak Level. The PLOUD Distribution Guidelines (EBU Tech Doc 3344 [5])
contains comprehensive coverage of the topic.
Strategies for loudness normalisation
Fig. 4 shows two approaches, mainly for production. The first is more relevant for the early stages
of the transition and it is perhaps especially useful to those who work on live programmes. Existing
meters, limiters and mixing practices are
retained and a level shift is done at the output of
1 2 the console (after the main meters) to achieve
the loudness Target Level of –23 LUFS. A loud-
ness meter is placed after the level shift to ena-
Change to ble the engineers to understand the exact
Keep levelling loudness level amount of shift (which initially is still a bit of
habit guesswork).
normalizaƟon
Using a loudness meter on past programmes of
the same genre gives good guidance as to
where the levels sit. Early experience at NDR,
ORF and RTBF has shown that it is certainly
Perform shiŌ No shiŌ needed possible for live mixes to fall into the ± 1 LU win-
dow permitted by R 128.
Figure 4 Those who work with files have an easier task as
Two principal working methods to achieve uniform the whole programme may be normalised
loudness in production and post-production
quickly and easily to –23 LUFS by means of a
level shift.
Those organizations or sections which can make the move to loudness metering straight away can
be expected to reap the benefit straight away. The greater dynamic range possible will be a wel-
come bonus for crowd noise in sports programmes – for example, enhancing the impact of a game
for the viewers and listeners.
EBU Tech Doc 3343 [3] gives practical guidelines for the new way of working with audio levels.
There are also basically two ways to achieve loudness normalisation for the consumer. One is the
actual normalisation of the source itself, so that the programmes are equally loud by design. The
other method is with the use of loudness metadata that describes how loud a programme is. For
the latter, the actual average programme loudness levels don’t need to be changed to a normalised
Abbreviations
dBFS dB relative to digital Full-Scale LU Loudness Unit
dBTP dB relative to digital Full Scale, measured with LUFS K-weighted Loudness Unit with reference to
a true peak meter digital Full Scale
FM Frequency Modulation
PML Permitted Maximum Level
LKFS Loudness, K-weighting, with reference to Full
Scale PPM Peak Programme Meter
LRA Loudness Range QPPM Quasi-Peak Programme Meter
EBU TECHNICAL REVIEW – 2010 Q3 5/9
F. Camerer
6. LOUDNESS
value and can still vary a lot. For those with up-
Loudness normalisation of to-date equipment, the normalisation can be per-
the source is recommended formed at the consumer’s end, using the individ-
ual loudness metadata values to gain-range the
in production because of programmes to the same replay level.
simplicity and potential Within the EBU R 128 loudness-levelling para-
quality gain. digm, the first solution is encouraged due to the
following advantages:
simplicity;
potential quality gain at the source.
The second solution is not forbidden (see also the Distribution Guidelines document – EBU Tech
Doc 3344 [5]), but having one single number (–23 LUFS) has great strength in spreading the loud-
ness-levelling concept, as it is easy to understand and act upon. And the active normalisation of the
source in a way “punishes” overcompressed signals and thus automatically encourages production
people to think about other, more dynamic and creative ways to make an impact with their pro-
gramme. In other words, the actual technical change of the source level through active normalisa-
tion to –23 LUFS has direct consequences on the artistic process – and in a positive way!
Nevertheless, it must be stated that both methods can complement each other: they are not to be
seen as opponents or a black-and-white view of the same issue. Both approaches are a part of
R 128 – but because of the advantages listed above, the normalisation of the source is recom-
mended.
Working with loudness meters
Until now, mixing with QPPMs and normalising the peaks have been done with reference to a per-
mitted maximum level (typically, –9 dBFS) and a peak limiter at that level provided a “safety ceiling”,
which could be hit as hard as desired – at the expense of less-engaging sound, of course.
In contrast, the loudness-levelling paradigm more resembles “floating in space” as the schematic
representation of a bar-graph meter in Fig. 5 shows. Fig. 6 shows how a software meter based on
a “needle” could look.
The EBU has deliberately not specified any graphical or user-interface details of a loudness meter,
but it has specified enhancements of the algorithm described in ITU-R BS.1770 and two scales:
EBU +9 Scale which ought to be suitable for most programmes;
EBU +18 Scale which may be used for programmes with a wide LRA.
Both scales can either display the relative Loudness Level in LU, or the absolute one in LUFS. The
meter manufacturers in the PLOUD Group have agreed to implement the “EBU mode” set of param-
Figure 5 Figure 6
A schematic representation of the two A schematic representation of an emulated
loudness scales (here in LU) as described loudness meter with a “bendy needle”
in EBU Tech 3341
EBU TECHNICAL REVIEW – 2010 Q3 6/9
F. Camerer
7. LOUDNESS
eters, to make sure their meters’ readings will be aligned. Many more manufacturers have adopted
“EBU mode” too, or are in the process of doing so.
An “EBU Mode” loudness meter as defined in EBU Tech Doc 3341 [6] offers three distinct time
scales:
Momentary Loudness (abbreviated “M”) – time window: 400ms
Short-term Loudness (abbreviated “S”) – time window: 3s
Integrated Loudness (abbreviated “I”) – from “start” to “stop”
The M and S time windows3 are to be used for the immediate levelling and mixing of audio signals.
Initial level setting may be performed best with the Momentary Loudness Meter, adjusting the level
of key or anchor elements (such as voice, music or sound effects) to be around the Target Level of
–23 LUFS. It is advisable to set levels with a bit of caution initially, as it is easier to gradually
increase the integrated loudness level during a mix than to decrease it. Usually, a slight increase in
the course of a programme is also more natural – and an initially “defensive” strategy leaves the
engineer room to manoeuvre in case of unexpected or unpredictable signals and events.
Once levels are set, the audio engineer can switch to mixing only by ear. Watching the Momentary
or Short-term Loudness, and an occasional glance at the Integrated Loudness value, should give
confirmation that the mix is within the tolerance allowed around the Target Level. With a numerical
readout of the I-value, with one decimal point precision, or a graphical display of similar resolution,
trends can be anticipated and the appropriate countermeasures taken.
To summarize ... improving the audio metering by replacing the PPM with the loudness meter is a
step closer to the best measurement tool – the human ear.
Loudness in the distribution chain
EBU Tech Doc 3344 [5] specifies the relevant settings and processing of the audio after it has left
the broadcast centre and takes the loudness paradigm all the way to the consumers’ equipment,
including set-top boxes and AV receivers. In doing so, it also encourages non-compliant broadcasts
to become compatible with EBU R 128. Digital transmissions, analogue transmissions, rebroadcast
transmissions, locally-inserted advertisements, newly-added services and much more are covered
in this comprehensive document.
Rather than measuring the loudness of individual programmes, the distribution company monitors a
service over 24 hours, with special care being taken when considering switched or shared services.
As well as the programmes, any loudness metadata provided with the digital services is monitored
so that the actual loudness level of the service may be compared with the stated level, both of which
should be –23 LUFS of course! Once a day, the data is analyzed and where a service is more than
0.9 LU from its Target Level (the exact value and method is still under discussion), a steering unit
applies a correction factor so that the long-term average of the audio remains at the Target Level
±1 LU.
Business consequences
Because the loudness-levelling paradigm affects all stages of an audio broadcast signal, from acqui-
sition to distribution and transmission and, because the ultimate goal is to harmonise audio loudness
levels within broadcast channels as well as between channels to achieve an equal universal loud-
3. “M” and “S” are commonly used in stereophony for “Mid” and “Side”. To distinguish the integration times
“Momentary” and “Short-term”, the versions “MLK” and “SLK” (as well as “ILK”) may be used. “LK” stands
for “Level, K-weighted”, and complies with the international naming standard ISO 80000-8.
EBU TECHNICAL REVIEW – 2010 Q3 7/9
F. Camerer
8. LOUDNESS
ness level for the benefit of the listeners, all audio professionals and audio-metering equipment
in all parts of the chain will be affected by this change.
For many, a major question will be whether all the existing QPPM meters will have to be replaced
and whether all the relevant personnel will need instruction to accommodate the new way of work-
ing. The answer is “In the long run, undoubtedly, yes”, but this transition needn’t happen all at once,
although at least some loudness meters should be brought into service at the earliest opportunity, to
sit alongside the QPPMs. The replacement of meters can happen in line with the technical refresh
cycles of other equipment, refurbishing of production facilities or as separate step-by-step projects
(whatever is feasible on a reasonably short time scale), and people can be trained at the appropriate
time.
Those responsible for the purchase of equipment should also be aware that safety limiters to pre-
vent over-modulation will have to be able to work in true-peak mode and they will need to be
adjusted to the appropriate maximum permitted true peak level, in production as well as at the out-
put of master control, at the distribution head-end and at the transmitter site.
Conclusions
EBU R 128 and the four supporting documents provide a means to end
the “Loudness War” – at last! The use of audio dynamics becomes a cre-
ative art again. There are still things to learn and people will take a while
to get used to the new ways of working, but it will be worth the effort.
Over 230 participants have joined the EBU PLOUD group (as at August
2010), the e-mail reflectors have shown a level of activity never seen
before and meter manufacturers have prepared units for display at
IBC 2010 before the specification was even published.
It is now time to put R 128 into action!
The EBU R 128 logo
Acknowledgements
The author especially wants to thank Ian Rudd, independent technology strategist and media con-
sultant, for his substantial editing contribution, as well as Frans de Jong, EBU, and Andrew Mason,
BBC R&D, for valuable input to this article.
References
[1] EBU Technical Recommendation R 128: Loudness normalisation and permitted maximum
level of audio signals
Florian Camerer is a senior sound engineer at ORF, the Austrian Broadcasting Corporation,
in Vienna. He started work in the area of production sound, expanding to sound editing and
mixing. His main field was documentaries, where he also developed a special interest in sur-
round sound. He mixed the first documentary of the ORF in 5.1 multichannel audio in 1995
and has been active in the area of surround sound ever since, helping ORF to become the first
European broadcaster to transmit a live 5.1 surround sound signal (New Year’s Concert,
2003).
In 2008, Mr Camerer asked the EBU to create a working group to study loudness issues which
led to the group PLOUD which he chairs. PLOUD is the biggest and most active EBU group
ever, which is reflected in the substantial output published.
Florian Camerer lectures on an international basis in surround sound and loudness matters.
EBU TECHNICAL REVIEW – 2010 Q3 8/9
F. Camerer
9. LOUDNESS
[2] ITU-R BS.1770: Algorithms to measure audio programme loudness and true-peak audio
level
[3] EBU Tech Doc 3343: Practical Guidelines for Production and Implementation in accord-
ance with EBU Technical Recommendation R 128 – publication awaited
[4] EBU Tech Doc 3342: Loudness Range: A descriptor to supplement loudness normalisa-
tion in accordance with EBU R 128
[5] EBU Tech Doc 3344: Practical Guidelines for Distribution of Programmes in accordance
with EBU R 128 – publication awaited
[6] EBU Tech Doc 3341: Loudness Metering: ‘EBU Mode’ metering to supplement loudness
normalisation in accordance with EBU R 128
This version: 6 September 2010
Published by the European Broadcasting Union, Geneva, Switzerland
ISSN: 1609-1469
Editeur Responsable: Lieven Vermaele
Editor: Mike Meyer
E-mail: tech@ebu.ch
The responsibility for views expressed in this article
rests solely with the author
EBU TECHNICAL REVIEW – 2010 Q3 9/9
F. Camerer