This document provides guidance on properly setting gain structure throughout an audio system. It explains that gain structure refers to setting various gain adjustments like mic preamps, faders, and output levels to work together properly. Improper gain setting can degrade sound quality, while proper gain structure maximizes the signal-to-noise ratio. The document outlines procedures for setting gains at each point in the system, from mixer inputs and outputs to any outboard gear and power amplifiers. The goal is to maximize the clean signal level at each stage to minimize noise contribution from subsequent stages.
This document provides descriptions and parameter information for various effect types available in the ZOOM multi-effects unit. It lists the name of each effect type, provides a brief explanation of its sound or modelled effect, and outlines the available parameters and their ranges for tweaking the effect. The parameters generally include settings for level, tone, gain, threshold, ratio and other effect-specific controls.
This document discusses different types of equalizers used in audio production including graphic, shelving, and parametric equalizers. Graphic equalizers have a fixed number of frequency bands that can each be boosted or cut using individual gain controls. Shelving equalizers broadly boost or cut the high and low frequencies, while parametric equalizers allow control over the central frequency, gain, and bandwidth (Q-factor) of an adjustable frequency band, providing the most precise equalization capabilities.
Mixers are electronic devices used to combine audio signals by routing and changing their level, tone, and dynamics. They allow adjustment of levels, equalization, effects, monitoring, and recording. Mixers come in various sizes from small portable units to large studio consoles. While intimidating for beginners due to many controls, mixers essentially have duplicated channel strips that make them easier to understand once you know how each channel works. Each channel strip contains gain, EQ, auxiliary sends, panning, and a level fader to control the signal flow and mix.
Pioneer AV Receivers 2012 - features of the LX SeriesPioneer Europe
The document discusses several new features of Pioneer AV receivers including Direct Energy HD Amplifiers using Direct Power FET technology, Phase Control technology to eliminate phase lag, Auto Phase Control Plus for Blu-ray and CD playback, and MCACC calibration system for acoustic optimization. It also covers Precision Quartz Lock System (PQLS) to prevent jitter during digital audio transmission, Sound Retriever technologies to improve compressed audio quality, and video processing technologies for improved picture quality.
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.
The document discusses the different sections of a mixing console, including the input section, routing, auxiliaries, equalization, channel path, and tape return path. The input section contains controls for the microphone and line inputs like gain, phantom power, filters. The routing matrix routes signals to tape or group faders. Auxiliaries are used to send signals to effects units or monitor mixes. The equalization section provides tone controls. The channel path and tape return path control signals from the mic input and tape, respectively.
The document provides an overview of Minleon RGB lighting products including:
- RGB bulbs come in different shapes with 3 color channels that can each display 256 intensity levels to create any color.
- Light strings have 3-4 wires to control individual or groups of bulbs. Controllers use faster XDMX protocol over traditional DMX.
- A variety of RGB bulbs, strips, tubes, and controllers are described along with their specifications and functions. Larger designs may require additional components for power distribution and data communication over long runs.
- Software like LightShow Pro is available to design light shows and synchronize them to music files. Various license levels provide control of different numbers of channels.
This document provides descriptions and parameter information for various effect types available in the ZOOM multi-effects unit. It lists the name of each effect type, provides a brief explanation of its sound or modelled effect, and outlines the available parameters and their ranges for tweaking the effect. The parameters generally include settings for level, tone, gain, threshold, ratio and other effect-specific controls.
This document discusses different types of equalizers used in audio production including graphic, shelving, and parametric equalizers. Graphic equalizers have a fixed number of frequency bands that can each be boosted or cut using individual gain controls. Shelving equalizers broadly boost or cut the high and low frequencies, while parametric equalizers allow control over the central frequency, gain, and bandwidth (Q-factor) of an adjustable frequency band, providing the most precise equalization capabilities.
Mixers are electronic devices used to combine audio signals by routing and changing their level, tone, and dynamics. They allow adjustment of levels, equalization, effects, monitoring, and recording. Mixers come in various sizes from small portable units to large studio consoles. While intimidating for beginners due to many controls, mixers essentially have duplicated channel strips that make them easier to understand once you know how each channel works. Each channel strip contains gain, EQ, auxiliary sends, panning, and a level fader to control the signal flow and mix.
Pioneer AV Receivers 2012 - features of the LX SeriesPioneer Europe
The document discusses several new features of Pioneer AV receivers including Direct Energy HD Amplifiers using Direct Power FET technology, Phase Control technology to eliminate phase lag, Auto Phase Control Plus for Blu-ray and CD playback, and MCACC calibration system for acoustic optimization. It also covers Precision Quartz Lock System (PQLS) to prevent jitter during digital audio transmission, Sound Retriever technologies to improve compressed audio quality, and video processing technologies for improved picture quality.
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.
The document discusses the different sections of a mixing console, including the input section, routing, auxiliaries, equalization, channel path, and tape return path. The input section contains controls for the microphone and line inputs like gain, phantom power, filters. The routing matrix routes signals to tape or group faders. Auxiliaries are used to send signals to effects units or monitor mixes. The equalization section provides tone controls. The channel path and tape return path control signals from the mic input and tape, respectively.
The document provides an overview of Minleon RGB lighting products including:
- RGB bulbs come in different shapes with 3 color channels that can each display 256 intensity levels to create any color.
- Light strings have 3-4 wires to control individual or groups of bulbs. Controllers use faster XDMX protocol over traditional DMX.
- A variety of RGB bulbs, strips, tubes, and controllers are described along with their specifications and functions. Larger designs may require additional components for power distribution and data communication over long runs.
- Software like LightShow Pro is available to design light shows and synchronize them to music files. Various license levels provide control of different numbers of channels.
The document discusses the history and implementation of automatic loudness control technology. It describes how CBS Laboratories developed the first loudness meter and controller in the 1960s to address loud commercials. Orban later implemented the CBS technology for use in television broadcast processing. The document examines how loudness control works, evaluating different processing methods. It analyzes loudness measurements to demonstrate how two-band and five-band compression with loudness control can regulate loudness over time better than unprocessed or compressed-only signals. The conclusions state that while imperfect, automatic loudness control effectively manages levels to prevent viewer annoyance.
This document discusses analog to digital conversion of sound. It explains that sound is captured through a microphone as changing electrical voltages representing amplitude and frequency. In analog recording, this is recorded continuously on tape. In digital recording, the voltage is sampled at regular time intervals and converted to binary digits representing amplitude and frequency values. Higher bit depths and sample rates allow for more accurate representation of sound. The document provides examples of how binary values represent decimal numbers and discusses setting up an audio interface and DAW for digital audio recording.
The document provides an overview of audio compression, including:
- Compression systems use a programme path and side-chain to reduce the dynamic range of a signal when its input level increases.
- Controls for compression include threshold, attack, release, ratio, knee, and stereo link. Threshold determines when compression is applied. Attack and release affect how quickly compression is applied and removed.
- Compression can be measured using meters showing gain reduction in dB. Compressors use different circuit types like VCA, FET, and opto.
Heli Newton from CMTO presented at the CBAA Conference 2018 on advanced audio editing and mixing techniques in Hindenburg. The presentation covered understanding terminology, identifying good and bad mixes, using compression and equalization plugins, and digital formatting and storage options. Examples were provided of bad and good mixes and the effects of compression and equalization were demonstrated. Formats for broadcasting and compression options were also discussed.
This document compares analog and digital signals. [1] Analog signals are continuous and take on a continuous range of values, represented by varying voltage levels. [2] They are characterized by amplitude, frequency, and phase. [3] Digital signals only have two states, on or off (1 or 0), making them better for digital computing but requiring more bandwidth than analog signals.
The document summarizes the signal flow through a typical analog mixing board channel strip. It begins with the input sources of microphones and line levels. It then describes the path the signal takes through a preamp, inserts, equalization, auxiliary sends, volume faders, panpots, and mute/solo switches before reaching the mix. The goal is to provide a basic overview of how sound is processed within a single channel of an analog mixing board.
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.
equalization in digital audio production graduation level education it is useful as the reference for bachelor of art students who choose mass communication as the main stream. a presentation from st.joseph's college
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.
Echo Chamber is a stereo reverb and delay plug-in effect, used to create psychoacoustic models to simulate sounds reflecting from surfaces in a room or space. Optionally a delay can be added to yield a spacious and open sound of a repeating, decaying echo to complete a sense of space and depth to a 'dry' input signal.
A highly tweakable, versatile, and inspiring solution for ambience effects, that produces a natural sounding room reverberation and delay effect giving a true room perception, from small rooms to large caverns as well as generates a doubling echo, slapback echo, ping-pong delay and analog tape delay. Offers multiple controls for modifying one or both channels to produce a rich array of time-based effects.
Available as plugin in VST and VST3 64 bit versions for Windows as well as in Audio Unit for macOS. These plug-ins are perfectly suited for any type of audio production when acoustic space simulation is needed from recording to post production in 64 bit platforms. Small rooms have a high percentage of early reflections (the first feedback from the closest objects) that can give more body to tracks. It is also good with acoustic guitars and voices. Larger rooms presets are better with strings, or wind instruments and synthesizer pads.
Features:
• Reverb and delay algorithms that delivers a rich reverberation and echoes by providing a spaciousness and depth to simulate the sound reflections from walls, floors and ceilings in an acoustically reflective environment.
• Flexibility to control Left and Right channels separately in Reverb and Delay units as well as in 'dry' signal output.
• Reverb unit works as a Stereo enhancer and mono-to-stereo creator, to produce a wide stereo image or stereoize a mono sound source. In Delay unit, improves the stereo image by adding a slight delay to one of the channels.
• Delay Time manual or synced to host (Tempo Sync BPM).
• 30 predefined space types, giving a virtually infinite number of possible shapes and sizes.
Preset Effects List:
01 • DEL - Analog Tape Delay
02 • DEL - Bucket Brigade Delay
03 • DEL - Crypt Echoes
04 • DEL - Doubling Echo
05 • DEL - Infinite Delay Machine
06 • DEL - Ping-Pong Delay
07 • DEL - Slapback Echo
08 • DEL - Sync Tube Tape Delay
09 • DEL - Tempo-Sync Delay
10 • DEL - Tube Driven Tape Echo
11 • REV - Amphitheater Reverb
12 • REV - Auditorium Reverb
13 • REV - Cathedral Reverb
14 • REV - Chamber Reverb
15 • REV - Hall Reverb
16 • REV - Opera Reverb
17 • REV - Plate Reverb
18 • REV - Room Reverb
19 • REV - Spring Reverb
20 • REV - Theater Reverb
21 • REV+DEL Ambience Reverb
22 • REV+DEL Arena Reverb
23 • REV+DEL Canyon Acoustics
24 • REV+DEL Catacomb Reverb
25 • REV+DEL Cave Reverb
26 • REV+DEL Church Reverb
27 • REV+DEL Cosmos Echo Panning
28 • REV+DEL Spatial Reverb
29 • REV+DEL Stadium Reverb
30 • REV+DEL Sync Bounced Delay
Mixing for games levels and more... jocelyn daoustMary Chan
Being part of the entertainment world, video games, gaming consoles and computers are everywhere. More and more people play on home theatre systems and believe that good sound plays a major role in their gaming experience.
The topic of mixing will be approached from technical, artistic and aesthetic standpoints.
Levels and measurement tools will be presented and explained.
The document discusses Digital Satellite News Gathering (DSNG) systems. DSNG systems allow news organizations to broadcast live from remote locations worldwide using satellite technology. There are two main types - mobile DSNG systems, where the equipment is mounted in a vehicle, and flyaway DSNG systems, where portable suitcase-sized equipment can be carried to locations. Key components include a 1.2m satellite antenna, indoor baseband units, upconverters, solid state power amplifiers, and backup generators. DSNG systems require careful link budget calculations to ensure sufficient signal strength over the satellite link.
Analogue an Digital Signals internet.pptmrmeredith
The document compares analogue and digital signals. It discusses how analogue signals are converted to digital signals for transmission, such as for digital television. It also explores the concepts of sampling rates for digital signals and how signals can become distorted at too low of a sampling rate through aliasing. The document notes that digital signals are more robust to noise than analogue signals since digital signals only consist of 1s and 0s and can be recovered exactly.
This document discusses dynamic processors and their use in audio production. Dynamic processors include compressors, limiters, expanders, and noise gates. They are used to control the dynamic range of a recording by making the volume more consistent or emphasizing certain parts. Key components of dynamic processors include the threshold, ratio, attack, and release settings. The threshold sets the input level at which processing begins, while the ratio determines how much the gain is reduced. The attack and release control how quickly processing engages and disengages.
This document provides an overview of how analog audio signals are converted to digital signals that can be read by computers and DAWs (digital audio workstations). It begins by defining digital audio as the process of encoding analog audio signals into binary information. This conversion process involves three main steps: sampling, quantization, and digital coding.
It then explains sampling as the process of taking discrete measurements of an analog signal's amplitude at regular intervals. These samples are converted to numerical values through quantization. Finally, digital coding translates the quantized values into binary code using a pre-established system of 0s and 1s. The document provides details on key concepts like sampling rate, quantization error, and bit depth throughout the conversion
This presentation covers noise performance of Continuous wave modulation systems; It explains modelling of white noise , noise figure of DSB-SC, SSB, AM, FM system
The document discusses different types of noise that affect communication systems, including thermal noise, shot noise, flicker noise, excess resistor noise, and popcorn noise. It provides details on thermal noise generation and its relation to temperature and resistance. The analysis section examines thermal noise in resistors in series and parallel and defines signal-to-noise ratio and noise factor. Additive white Gaussian noise is described as noise that is additive, has a constant spectral density (white), and has a Gaussian amplitude distribution.
This document provides an overview of analog and digital signals, periodic signals, digital signals, and transmission impairment. It discusses topics such as:
- Analog signals are continuous while digital signals have discrete states
- Periodic signals can be simple or composite, with a composite made of multiple sine waves
- Digital signals have a bit rate and bandwidth requirement for transmission
- Transmission is impaired by attenuation, distortion, and noise, which can be measured by signal-to-noise ratio
- Data rate limits depend on bandwidth, signal levels, and channel noise as defined by Nyquist rate and Shannon capacity.
The document provides an overview of setting up a basic live sound system, including:
1) Describing the signal flow from mixing console through graphic equalizers, crossover, power amplifiers, and main speakers.
2) Explaining how to set up monitor speakers using auxiliary outputs from the mixing console.
3) Providing details on setting gain and equalization on each channel strip of the mixing console.
This document provides a parameter list and description for the AMEK Mastering Compressor plugin. It includes over 30 parameters organized into sections for general controls, channel processing, sidechain filtering and monitoring, and meters. The parameters allow for detailed customization of compression, EQ, stereo imaging, and more. Key features include soft and hard knee compression, dual RMS detectors, mid/side processing, and full sidechain control.
The document discusses the history and implementation of automatic loudness control technology. It describes how CBS Laboratories developed the first loudness meter and controller in the 1960s to address loud commercials. Orban later implemented the CBS technology for use in television broadcast processing. The document examines how loudness control works, evaluating different processing methods. It analyzes loudness measurements to demonstrate how two-band and five-band compression with loudness control can regulate loudness over time better than unprocessed or compressed-only signals. The conclusions state that while imperfect, automatic loudness control effectively manages levels to prevent viewer annoyance.
This document discusses analog to digital conversion of sound. It explains that sound is captured through a microphone as changing electrical voltages representing amplitude and frequency. In analog recording, this is recorded continuously on tape. In digital recording, the voltage is sampled at regular time intervals and converted to binary digits representing amplitude and frequency values. Higher bit depths and sample rates allow for more accurate representation of sound. The document provides examples of how binary values represent decimal numbers and discusses setting up an audio interface and DAW for digital audio recording.
The document provides an overview of audio compression, including:
- Compression systems use a programme path and side-chain to reduce the dynamic range of a signal when its input level increases.
- Controls for compression include threshold, attack, release, ratio, knee, and stereo link. Threshold determines when compression is applied. Attack and release affect how quickly compression is applied and removed.
- Compression can be measured using meters showing gain reduction in dB. Compressors use different circuit types like VCA, FET, and opto.
Heli Newton from CMTO presented at the CBAA Conference 2018 on advanced audio editing and mixing techniques in Hindenburg. The presentation covered understanding terminology, identifying good and bad mixes, using compression and equalization plugins, and digital formatting and storage options. Examples were provided of bad and good mixes and the effects of compression and equalization were demonstrated. Formats for broadcasting and compression options were also discussed.
This document compares analog and digital signals. [1] Analog signals are continuous and take on a continuous range of values, represented by varying voltage levels. [2] They are characterized by amplitude, frequency, and phase. [3] Digital signals only have two states, on or off (1 or 0), making them better for digital computing but requiring more bandwidth than analog signals.
The document summarizes the signal flow through a typical analog mixing board channel strip. It begins with the input sources of microphones and line levels. It then describes the path the signal takes through a preamp, inserts, equalization, auxiliary sends, volume faders, panpots, and mute/solo switches before reaching the mix. The goal is to provide a basic overview of how sound is processed within a single channel of an analog mixing board.
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.
equalization in digital audio production graduation level education it is useful as the reference for bachelor of art students who choose mass communication as the main stream. a presentation from st.joseph's college
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.
Echo Chamber is a stereo reverb and delay plug-in effect, used to create psychoacoustic models to simulate sounds reflecting from surfaces in a room or space. Optionally a delay can be added to yield a spacious and open sound of a repeating, decaying echo to complete a sense of space and depth to a 'dry' input signal.
A highly tweakable, versatile, and inspiring solution for ambience effects, that produces a natural sounding room reverberation and delay effect giving a true room perception, from small rooms to large caverns as well as generates a doubling echo, slapback echo, ping-pong delay and analog tape delay. Offers multiple controls for modifying one or both channels to produce a rich array of time-based effects.
Available as plugin in VST and VST3 64 bit versions for Windows as well as in Audio Unit for macOS. These plug-ins are perfectly suited for any type of audio production when acoustic space simulation is needed from recording to post production in 64 bit platforms. Small rooms have a high percentage of early reflections (the first feedback from the closest objects) that can give more body to tracks. It is also good with acoustic guitars and voices. Larger rooms presets are better with strings, or wind instruments and synthesizer pads.
Features:
• Reverb and delay algorithms that delivers a rich reverberation and echoes by providing a spaciousness and depth to simulate the sound reflections from walls, floors and ceilings in an acoustically reflective environment.
• Flexibility to control Left and Right channels separately in Reverb and Delay units as well as in 'dry' signal output.
• Reverb unit works as a Stereo enhancer and mono-to-stereo creator, to produce a wide stereo image or stereoize a mono sound source. In Delay unit, improves the stereo image by adding a slight delay to one of the channels.
• Delay Time manual or synced to host (Tempo Sync BPM).
• 30 predefined space types, giving a virtually infinite number of possible shapes and sizes.
Preset Effects List:
01 • DEL - Analog Tape Delay
02 • DEL - Bucket Brigade Delay
03 • DEL - Crypt Echoes
04 • DEL - Doubling Echo
05 • DEL - Infinite Delay Machine
06 • DEL - Ping-Pong Delay
07 • DEL - Slapback Echo
08 • DEL - Sync Tube Tape Delay
09 • DEL - Tempo-Sync Delay
10 • DEL - Tube Driven Tape Echo
11 • REV - Amphitheater Reverb
12 • REV - Auditorium Reverb
13 • REV - Cathedral Reverb
14 • REV - Chamber Reverb
15 • REV - Hall Reverb
16 • REV - Opera Reverb
17 • REV - Plate Reverb
18 • REV - Room Reverb
19 • REV - Spring Reverb
20 • REV - Theater Reverb
21 • REV+DEL Ambience Reverb
22 • REV+DEL Arena Reverb
23 • REV+DEL Canyon Acoustics
24 • REV+DEL Catacomb Reverb
25 • REV+DEL Cave Reverb
26 • REV+DEL Church Reverb
27 • REV+DEL Cosmos Echo Panning
28 • REV+DEL Spatial Reverb
29 • REV+DEL Stadium Reverb
30 • REV+DEL Sync Bounced Delay
Mixing for games levels and more... jocelyn daoustMary Chan
Being part of the entertainment world, video games, gaming consoles and computers are everywhere. More and more people play on home theatre systems and believe that good sound plays a major role in their gaming experience.
The topic of mixing will be approached from technical, artistic and aesthetic standpoints.
Levels and measurement tools will be presented and explained.
The document discusses Digital Satellite News Gathering (DSNG) systems. DSNG systems allow news organizations to broadcast live from remote locations worldwide using satellite technology. There are two main types - mobile DSNG systems, where the equipment is mounted in a vehicle, and flyaway DSNG systems, where portable suitcase-sized equipment can be carried to locations. Key components include a 1.2m satellite antenna, indoor baseband units, upconverters, solid state power amplifiers, and backup generators. DSNG systems require careful link budget calculations to ensure sufficient signal strength over the satellite link.
Analogue an Digital Signals internet.pptmrmeredith
The document compares analogue and digital signals. It discusses how analogue signals are converted to digital signals for transmission, such as for digital television. It also explores the concepts of sampling rates for digital signals and how signals can become distorted at too low of a sampling rate through aliasing. The document notes that digital signals are more robust to noise than analogue signals since digital signals only consist of 1s and 0s and can be recovered exactly.
This document discusses dynamic processors and their use in audio production. Dynamic processors include compressors, limiters, expanders, and noise gates. They are used to control the dynamic range of a recording by making the volume more consistent or emphasizing certain parts. Key components of dynamic processors include the threshold, ratio, attack, and release settings. The threshold sets the input level at which processing begins, while the ratio determines how much the gain is reduced. The attack and release control how quickly processing engages and disengages.
This document provides an overview of how analog audio signals are converted to digital signals that can be read by computers and DAWs (digital audio workstations). It begins by defining digital audio as the process of encoding analog audio signals into binary information. This conversion process involves three main steps: sampling, quantization, and digital coding.
It then explains sampling as the process of taking discrete measurements of an analog signal's amplitude at regular intervals. These samples are converted to numerical values through quantization. Finally, digital coding translates the quantized values into binary code using a pre-established system of 0s and 1s. The document provides details on key concepts like sampling rate, quantization error, and bit depth throughout the conversion
This presentation covers noise performance of Continuous wave modulation systems; It explains modelling of white noise , noise figure of DSB-SC, SSB, AM, FM system
The document discusses different types of noise that affect communication systems, including thermal noise, shot noise, flicker noise, excess resistor noise, and popcorn noise. It provides details on thermal noise generation and its relation to temperature and resistance. The analysis section examines thermal noise in resistors in series and parallel and defines signal-to-noise ratio and noise factor. Additive white Gaussian noise is described as noise that is additive, has a constant spectral density (white), and has a Gaussian amplitude distribution.
This document provides an overview of analog and digital signals, periodic signals, digital signals, and transmission impairment. It discusses topics such as:
- Analog signals are continuous while digital signals have discrete states
- Periodic signals can be simple or composite, with a composite made of multiple sine waves
- Digital signals have a bit rate and bandwidth requirement for transmission
- Transmission is impaired by attenuation, distortion, and noise, which can be measured by signal-to-noise ratio
- Data rate limits depend on bandwidth, signal levels, and channel noise as defined by Nyquist rate and Shannon capacity.
The document provides an overview of setting up a basic live sound system, including:
1) Describing the signal flow from mixing console through graphic equalizers, crossover, power amplifiers, and main speakers.
2) Explaining how to set up monitor speakers using auxiliary outputs from the mixing console.
3) Providing details on setting gain and equalization on each channel strip of the mixing console.
This document provides a parameter list and description for the AMEK Mastering Compressor plugin. It includes over 30 parameters organized into sections for general controls, channel processing, sidechain filtering and monitoring, and meters. The parameters allow for detailed customization of compression, EQ, stereo imaging, and more. Key features include soft and hard knee compression, dual RMS detectors, mid/side processing, and full sidechain control.
Sound Engineering introduction to mixersYEducation
Basics of sound Engineering introduction to mixers and how to operate them
What Is Sound Engineering?
Different Types of Sound Engineering
How to Become A Sound Engineer?
live and studio engineering
The document describes 70V paging systems and their components. 70V systems use a centralized amplifier with a 70V output to power speakers connected via long runs of wire. This high voltage, low current design minimizes power loss over the wires. The document discusses the benefits of 70V systems and explains their basic design and components, such as the centralized amplifier, speakers, taps, and interface devices. It also covers an alternative self-amplified paging system design that uses individual amplifiers built into each speaker powered by a 24V DC supply.
The document provides assembly and use instructions for the RelaiXed, a high-end pre-amplifier designed for balanced audio connections. Key features include a 64-step relay-based volume attenuator, infrared remote control, and separate power supplies per audio channel. Assembly requires carefully sourcing quality components like gold-plated PCBs, LM4562 op-amps, and Dale RN60-series resistors to achieve the best sound quality.
The document summarizes key functions and components of a mixing console:
1) A mixing console has three main functions - amplification, mixing, and routing. It amplifies audio signals, allows adjustment of signal levels during mixing, and routes signals to different destinations using buses.
2) Proper gain setting is important for clear audio - too low results in noise, too high causes distortion. Clip indicators help monitor levels and prevent distortion.
3) Mixing involves adjusting relative signal levels using faders or potentiometers to cut or reduce levels. Meters help calibrate and monitor levels across the recording chain.
This document defines many common audio and electronic terms used in sound mixing and recording. It provides brief definitions and explanations of terms like EQ, compressor, delay, reverb, DAW, gain staging, and more. It also recommends several textbooks for readers to reference if they want more detailed information on audio engineering terms and concepts.
This document provides instructions for the TOPP PRO TPA GIG7.800-15 PACK audio equipment package. The package includes a TPM7.800 powered mixer with a 230W amplifier and 24-bit multi-effects processor, and two TPS 115 NEO passive speaker cabinets. The mixer provides 10 input channels, effects, equalization, and output options. The speaker cabinets are designed for quality and portability. The document reviews the features and controls of the mixer and advises reading the manual to take advantage of the full functionality.
Sound Engineering, What Does It Involve?.pptxDerek896886
This document provides an overview of a sound engineering course, including:
- An introduction to the process of making a record from writing songs to mixing and mastering in the studio.
- An explanation of the roles of people involved like producers, engineers, and assistants.
- A basic explanation of studio signal flow from microphones and instruments to the DAW via analog to digital conversion.
- Descriptions of monitoring options like near field, mid field, and main speakers.
- An exercise for students to work together to program, record, and mix an original track.
The document is a user manual for the PS 260 dual channel audio interface. It describes the product's general features, including two separate audio interfaces that allow connection between an ASL partyline intercom system and external 4-wire audio equipment. The user manual provides details on installation, front panel controls, rear panel connections, and setup instructions.
The document is a user manual for the PS 260 dual channel audio interface. It describes the product's general features, including two separate audio interfaces that allow connection between an ASL partyline intercom system and external 4-wire audio equipment. The user manual provides details on installation, front panel controls, rear panel connections, and setup instructions.
The document is a user manual for the PS 260 dual channel audio interface. It describes the product's general features, including two separate audio interfaces that allow connection between an ASL partyline intercom system and external 4-wire audio equipment. The user manual provides details on installation, front panel controls, rear panel connections, and setup instructions.
This document discusses the signal flow through audio consoles. It explains that understanding this signal flow is critical for sound engineers to troubleshoot problems. It then describes the basic signal path through analog consoles from the microphone or line input, through the channel strip including inputs, equalization, pans, and faders, to the master output. It also discusses digital consoles and how their signal flow is similar but routing is controlled digitally rather than with individual channel controls.
Five Things about PA I wish I had learnt years agoMusicademy
Seminar notes from Musicademy training day with content by SFL Ltd church PA.
An excellent introduction to the basics of PA.
Sign up here for over 40 free lessons from Musicademy including one from our Sound Tech Training DVD
http://bit.ly/12S4iPP
The Nady DKW-3 is a wireless microphone system that offers clear audio transmission for speaking engagements without cables. It features a handheld transmitter and receiver with a range of 150 feet and controls like power switches and volume. The system provides professional audio quality while being affordable and easy to use.
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La ingeniería de sonido en vivo ha evolucionado desde los teatros griegos y el Coliseo Romano hasta los grandes conciertos del siglo XX. En la actualidad, los sistemas de sonido utilizan arrays de altavoces para proyectar el sonido de manera direccional al público, y las consolas digitales permiten mezclar múltiples canales de entrada. Los ingenieros de sonido se encargan de tareas como la mezcla en el FOH, los monitores y la grabación para asegurar una experiencia óptima.
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Este documento describe la evolución de la grabación musical desde el fonógrafo de Edison hasta Spotify, incluyendo el uso inicial de cilindros y discos, la grabación en bloque y multipista, el desarrollo de grabadoras y estudios de grabación profesionales, y el cambio hacia formatos digitales y de streaming.
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1. 205Compact Mixer Reference Guide
MIXER TIPS: CHAPTER 4
Gain Structure - Setting the System Levels
One of the most important things you can do to
make an audio system sound good is to set up its gain
structure properly. Conversely, improper gain setting
throughout the system can really make it sound bad.
Gain Structure is the term we use for the collec-
tion of various gain adjustments throughout the
system – the mic preamp, the fader, the main mix
output level, the input gain of a power amplifier or
recorder, and so on. Setting all of those gain/level
controls to work together properly isn’t difficult,
but so often it gets ignored. You may run across the
term Gain Staging. That’s the process of setting
gains throughout the system to achieve proper gain
structure.
We hope that by now you’ve at least tattooed the
Level-Setting Procedure on the back of your hand.
Since the TRIM controls the gain at the point where
the sound enters your system, its proper adjustment
is fundamental to proper gain staging.
There are many ways of establishing optimum gain
structure. Once you understand what you need to
accomplish, gain staging will become second nature
whenever you connect or operate new equipment.
To understand gain structure and gain staging, you
need to understand a couple of closely related terms
that we’ve sprinkled throughout this book – dynamic
range and headroom.
DynamicRange
Dynamic range is the ratio, expressed in dB (dog
biscuits), of the level of the loudest undistorted
signal to that of the quietest audible signal. Dynamic
range can apply to a single piece of equipment or to a
complete system.
For electronic equipment, the maximum output
is ultimately determined by its power supply. If the
power supply provides ±15 volts to the integrated
circuit opamps, the maximum possible peak-to-peak
signal voltage is 30 volts. It will never be 31 volts
— you can’t generate what’s not there, at least not
without adding more components.
The noise floor limits the lowest audible signal
we can use in the system. In general, the better the
design, the lower the noise floor. Reducing the noise
floor increases the available dynamic range.
Typical professional audio equipment like your
Mackie mixer can put out a maximum level of +22
to +28 dBu depending on the model and the output
jack. Noise floor is a little harder to measure (there
are several legitimate methods), but it’s usually in
the ballpark of –85 dBu with a bunch of channels
assigned and set to unity gain. This gives a maximum
possible dynamic range of 113 dB – pretty impressive
considering that the dynamic range of human hear-
ing (threshold of hearing to threshold of discomfort)
is around 120 dB.
It’s convenient that electronic system noise is
usually considerably lower than ambient noise such
as traffic and air conditioning. Usable dynamic range
is rarely greater than 100 dB in a quiet studio and
may be as little as little as 20 dB at a concert with an
audience of 20,000 screaming teenage girls. (More
civilized concert venues usually allow for dynamic
range on the order of 55-70 dB.)
Headroom
Headroom is the ratio of the largest possible un-
distorted signal to the average signal level. Average
level is subject to some interpretation depending on
whether you’re the sound reinforcement engineer at
a concert or the promoter asking you to turn it up. At
a loud concert, the average sound pressure level may
be very close to the maximum possible level (very
little headroom). If you’re writing the spec sheet of
a console, however, you want to be able to show the
greatest possible headroom above the nominal level.
When it comes to electronics, the average level is
generally considered to be the equipment’s nominal
operating level. If the nominal level is +4 dBu, our
mixer’s Main outputs, with their +28 dBu maximum
output capability, will allow 24 dB of headroom.
Since there’s little you can do about the system’s
dynamic range once you’ve chosen your equipment
(short of turning off the air conditioner or gagging
the screaming teenage girls), all you need to do in
order to assure undistorted sound is to provide suf-
ficient headroom. Sounds simple, but how much is
sufficient?
Crest Factor
Crest factor is the ratio of the peak to the RMS
(average) value of the signal. It’s a micro-measure-
ment though, not a long term average like sound
pressure level or a sine wave voltage measurement.
Crest factor is measured within the waveform cycle.
2. 206 Compact Mixer Reference Guide
MIXER TIPS: CHAPTER 4
As a simple example, the RMS value of a sine wave
is 0.707 times the peak value. So its crest factor is
1/0.707 = 1.4, or 3 dB.
Empirical studies have shown that typical pop
music has a crest factor in the range of 4 to 10, which
translates to 12 to 20 dB. This means that we need to
be able to provide 12 to 20 dB of headroom for peaks
over the average level in order to avoid clipping. This
is a requirement for every link in the chain.
TakeitFromTheTop–SettingSystem
Levels
Once you have the whole system hooked up,
whether it’s your studio control room or a concert
sound reinforcement system, verify that you have
all the proper connections and it will pass audio. Do
this before you start setting gains so you don’t get
sidetracked with troubleshooting. Put on a CD, make
sure music gets to the output(s), and then listen
for hum and buzzes that will eat your noise floor for
lunch. Once you’re sure the system is passing signals
cleanly and properly, you can start gain setting.
Pre-flight Check List
• Turn down all the power amplifier input level
controls
• Turn the power amplifiers off (so you can run
test signals through the system all the way to
the amplifiers without driving yourself bon-
kers)
• Set all gain/level controls to their minimum
gain positions.
• Bypass or zero out any equalization, both chan-
nel EQ and overall EQ on the outputs.
• Bypass any compressors or limiters, or set
their threshold all the way up so they don’t
compress.
ConsoleGainSettings
All mixing consoles consist of a mic/line preamp
stage, equalization, a channel fader, and channel
routing controls (PAN pots and ASSIGN switches).
There may or may not be a submaster stage depend-
ing on the model and whether a channel is assigned
to a SUB or directly to the MAIN outputs. Finally,
all channels are mixed together to various outputs
(MAIN, AUX, Control Room, etc.) most of which have
their own level control.
To set gain structure properly, you want to maxi-
mize the signal-to-noise (S/N) ratio. This requires
some thought and care, as each stage along the path
contributes some noise as well as gain. The pessimist
looks at this and sees that each stage contributes to
the degradation of S/N ratio – and, you know, he’s
right. But the amount of noise a stage contributes is
fixed, so the higher the signal level is at its input, the
better the S/N ratio at the output.
It’s generally good practice (and this is as much a
designer’s issue as an operator’s) to bring the input
signal up to the desired operating level (say +4 dBu)
as early in the signal chain as possible. If you need
to amplify a microphone by 60 dB in order to get to
the final desired output level, it’s best not to do it in
steps – 20 dB at the preamp, 20 dB in the equalizer
section, and another 20 dB at the output. You want
to put as much of that 60 dB gain as possible right
at the preamp stage and run everything else close to
unity gain. That’s why we have the famous Mackie
Level-Setting Procedure and those little “U” marks on
just about every control.
By soloing a channel, you send its preamp output
to the meter. Adjusting the TRIM control for a 0 VU
meter reading on peaks sets the preamp gain so that
you’ll have at least 20 dB of headroom before encoun-
tering clipping in the preamp.
This may seem like plenty, but remember, you’ll be
adding up a bunch of channels by the time you get to
the main output bus.
Audio signals add as the square root of
the sum of the square of their amplitudes.
Since 0 VU on the meter represents a level
of 0.775 volts, get 24 channels cranking all at once
and you’ll get 3.8 volts (which translates to around
+14 dBu) out of the mixer. That’s a theoretical case
with 0 VU sine waves on every channel. But with real
music, at any instant in time some channels will be
peaking higher, others lower, so it all about averages
out.
With the input gain set “by the meter,” as long as
the rest of the console is at unity gain, with the 20 dB
of preamp headroom, that extra-loud scream won’t
clip internally in the mixer. With all channels crank-
ing and the output fader set to its unity position,
the meters should be running in the ballpark of +15
VU. Since there’s around 10 dB (depending on the
model) of headroom at the output at this level, under
any reasonable conditions, the mixer won’t be the
source of clipping in a system.
3. 207Compact Mixer Reference Guide
MIXER TIPS: CHAPTER 4
The reason why it’s so important to
set the gain properly at the input is
because the mixer’s VU meters don’t
have very good resolution up near
the critical top of the range. There’s
a –10 LED which you’ll probably be
hitting frequently, but the next one
up from there is at the maximum
output level. So the actual level
between pretty hot but still safe and
clipping is ambiguous.
OutboardGearLevels
Many outboard units, including power amplifiers,
are designed so they operate at “unity gain,” and
have no level controls at all. Others have input level
controls, and electronic crossovers, equalizers, com-
pressors, or other in-line signal processors often have
both input and output level controls.
In a perfect world, having no input level control
would be OK. The console would add all the gain
necessary and provide all the control. Often, however,
particularly in live-sound situations, it’s necessary to
turn up the output level of the console to the point
where it’s at risk of clipping in order to get sufficient
volume at the speakers. This is not a good situation
and indicates that the power amplifier isn’t properly
matched to the system.
It’s not uncommon for a loud sound
reinforcement system to be running very
close to clipping most of the time. Very
brief clipping is usually not noticeable or harmful,
but sustained clipping sounds awful and can damage
your speakers.
Setting Outboard Levels
Begin by turning the power amplifiers
off or setting their input gain controls to
minimum. Otherwise, you’ll be in for a lot
of noise.
Set the console so that it’s producing maximum
output level. Play a CD or use an external tone
generator. After assuring that the input isn’t clip-
ping (that ol’ Level-Setting Procedure again), set the
channel fader (or AUX SEND control if you’re setting
up something connected to an AUX output) to Unity
gain, then raise the MASTER fader (and channel
fader, if necessary) to bring the output up to the
maximum level before clipping as indicated by the
VU meter. This is the hottest signal your mixer can
send to that output.
If the mixer is connected directly to a power
amplifier, skip to the paragraph about setting power
amplifier gain. If there’s a device or two in between,
you’ll need to set those levels before getting to the
power amp. Keep reading from here.
When going into a device that has only an input
level control, set its input control to the point where
that device is putting out its maximum level before
clipping. If it has a clip indicator or level meter, rely
on that to tell you when you’re there. If not, crack the
gain on the power amplifier (hopefully there’s one)
just enough to hear and listen for clipping. It’ll be
pretty obvious.
If the in-line device has both an input gain and an
output level control, it’s important to set the input
gain first. Do this by turning the output level up just
enough to hear the signal, set the input gain to the
point just before clipping, then finally set the output
level so the output is just below clipping.
See what’s happening here? We’re squeezing
the last clean decibel out of the console, and then,
with the knowledge that we’ll never be driving it
any harder than the maximum console output level,
squeezing the last clean decibel out of the next unit
in line, and so on. By feeding the outboard chain with
the maximum clean console output level, we’re as-
suring that we’ll be getting the largest gain boost up
front where it belongs, and the rest of the system will
be loafing and not amplifying noise.
Setting Power Amplifier Gain
First, understand that the input gain controls on
your power amplifier are sensitivity controls. They
have nothing to do with the amount of power the am-
plifier can produce. That’s fixed by the design. What
the control (or the manufacturer, if there’s no input
gain control) determines is how much input voltage
is required in order for the amplifier to produce its
full rated power.
With the control turned up full, a high-sensitivity
amplifier might produce full power with an input
level as low as –20 dBu, or that might be point at
which it produces full power at the nominal input
level of +4 dBu. Turning the input level control down
from its maximum sensitivity position increases the
signal level required to get maximum power out of
the amplifier. Once you understand that, it’s pretty
simple to set the amplifier gain properly. You want
the power amplifier to put out full power when the
rest of the system is fully cranked.
LEFT RIGHT
28
CLIP
10
7
4
2
2
0
4
7
10
20
30
40
OPERATING LEVEL
0dB = 0dBu
4. 208 Compact Mixer Reference Guide
MIXER TIPS: CHAPTER 4
Setting the amplifier sensitivity too low means
you’ll never reach full power but, in trying, you can
send a pretty darn loud clipped signal (which will be
cleanly amplified) to the speakers and your audi-
ence. More sensitivity than necessary means that you
can’t take advantage of all the headroom in the mixer
because you’ll run out of headroom in the power
amplifier first.
To set the amplifier input sensitivity properly:
• Hold your ears and warn everyone else within
hearing range. You’ve been sparing your ears
for a while, but now it’s time to make noise.
• Turn the amplifier’s input gain controls all the
way down. If the power amplifier is off, turn it
on.
• Crank up your music or test tone to maxi-
mum level below clipping as indicated on the
console’s VU meters.
• Turn up the amplifier’s input gain control until
you can just hear clipping or the amplifier’s
clip indicator just goes on, then back it off a
little bit. This is the proper gain setting. Leave
it there.
If the volume is too loud (Hah! In your dreams!),
it’s fine now to turn down the master level at the
console. You don’t have to turn on all those meter
lights just because you paid for them. If it’s not loud
enough, it isn’t going to get any louder – the amplifier
is undersized, the speakers aren’t efficient enough, or
they’re placed incorrectly.
InTheStudio
The same principles of gain structure apply when
setting up a recording chain. You don’t want your
recorder to run out of headroom before the mixer
output that’s feeding it. Otherwise you won’t be feed-
ing the recorder a signal with the best signal-to-noise
ratio. Conversely, you want your recorder’s input
sensitivity to be high enough so that you can reach
maximum recording level without the mixer clipping.
Many recorders today don’t have input level
controls, so to a certain extent you’re at the mercy of
the designers. Fortunately, you’re usually safe if you
connect inputs and outputs with matching nominal
operating levels.
MismatchedComponents
Sometimes the best theoretical setup doesn’t work
in practice and you need to compensate somewhere.
Suppose you’ve set up your system for maximum
headroom, and when the console meters are just
hitting 0 VU, the sound is so loud that people are
leaving. OK, so you turn the console main mix level
down and they come back in when your console me-
ters are peaking around –20 VU. But now the meters
on the recorder that you have connected to the main
outputs are barely moving and you’re concerned that
you’ll get a noisy recording. And on top of that, the
console faders are so close to the bottom that you
don’t have very much working range to get a good
mix.
The thing to do in this case is to reduce the input
sensitivity of the power amplifiers so that with the
mixer running at its optimum level (around 0 VU on
its meters), you’re getting the right volume level in
the house. True, you’re compromising the gain struc-
ture a bit, but you do what you have to do. It’s far less
destructive to run the last stage in the chain at lower
than optimum gain than the first stage.