This document discusses multi-track audio workflows for cinema and video production. It begins by outlining the evolution from monaural to two-track to four-track recording. For two-track recording, it describes using one track for a live mix and the other for iso tracks or effects, or splitting mics between tracks. For four-track recording, it discusses options like recording iso tracks or a live mix plus iso tracks. It also covers techniques for 8+ track recordings like recording iso tracks and an attenuated dual mono mixdown. The document provides guidance on mic selection, placement, and track assignments.
This document provides an overview of sound production equipment, including sound desks and cables. It describes the main components and functions of a sound desk, such as its input/output sockets, channels, faders to control volume, gain to control depth of volume, and peak monitor to check speaker volume levels. It also covers different types of cables used, including XLR, audio jack, and speaker cables. Additionally, it compares active speakers with built-in amplifiers to passive speakers that require a separate amplifier.
The document provides instructions for setting up a voice over recording studio. It lists 8 steps to remember as a technical team, including soundproofing the room, managing noise levels, arranging your work station, and equipment needed. Key equipment includes a laptop, audio mixer, microphone, music stand, and monitor speakers. The document then describes how to set up the audio recording equipment, explaining the common controls on a mixing console like channel inputs, gain control, equalization controls, and master volume. It emphasizes never giving up and having confidence in your work.
This document provides information about technical stage operations related to sound equipment. It includes diagrams and explanations of how a basic PA system works, including sound sources, the sound desk, amplifiers, and active or passive speakers. It describes the inputs, outputs, and controls of a sample sound desk. It also covers topics like equalizers, reverb, delay, auxiliary controls, and different types of input cables.
This document contains information about various audio equipment including microphones, audio interfaces, studio monitors, headphones, and a MIDI keyboard controller. It provides specifications for the Line 6 TonePort UX1 and M-Audio Fast Track audio interfaces, Shure SM58 microphone, Behringer C-1 and C-2 condenser microphones, M-Audio Axiom 49 and KeyRig49 MIDI keyboard controllers, Sony earbuds, Sennheiser headphones, Xiaomi earphones, Samson MediaOne studio monitors, and Behringer Tube Ultragain Mic100 microphone preamplifier.
There are two types of speakers: active speakers, which contain an internal amplifier and only require a power source and sound input, and passive speakers, which do not contain an amplifier and require an external amplifier and connections to a sound source. Different cables are used to connect speakers to audio equipment, including DMX cables between speakers and sound desks, phono cables between sound sources and mixing desks, and various jack and power cables. A mixing desk controls audio by adjusting properties like volume, EQ levels, and panning for each input source and sending the processed audio signals to left and right speakers.
This document discusses frequency and its applications in equalization (EQ) and sound synthesis. It defines frequency as the number of cycles per second and describes how frequency is perceived as pitch. It discusses units of measurement like Hertz and how humans can typically hear frequencies between 20-20,000 Hz. It then covers topics like the frequency ranges of instruments, EQ, octaves, EQ bands, different types of EQ like parametric EQ, shelf EQ, and notch filters. It includes assignments to experiment with EQ in iTunes and record vocals to sound like they are coming from a telephone.
A compact, four channel field recorder capable of professional quality recordings. It supports up to 24-bit/192kHz sample rates and large capacity SDHC cards. Built-in effects, filters, and limiters allow for flexible audio editing and cleanup. Connectivity includes XLR, 1/4 inch, and digital inputs/outputs. Up to eight channels can be recorded by linking two units. The lightweight design runs on AA batteries or USB power, making it a portable solution for location sound.
This document discusses multi-track audio workflows for cinema and video production. It begins by outlining the evolution from monaural to two-track to four-track recording. For two-track recording, it describes using one track for a live mix and the other for iso tracks or effects, or splitting mics between tracks. For four-track recording, it discusses options like recording iso tracks or a live mix plus iso tracks. It also covers techniques for 8+ track recordings like recording iso tracks and an attenuated dual mono mixdown. The document provides guidance on mic selection, placement, and track assignments.
This document provides an overview of sound production equipment, including sound desks and cables. It describes the main components and functions of a sound desk, such as its input/output sockets, channels, faders to control volume, gain to control depth of volume, and peak monitor to check speaker volume levels. It also covers different types of cables used, including XLR, audio jack, and speaker cables. Additionally, it compares active speakers with built-in amplifiers to passive speakers that require a separate amplifier.
The document provides instructions for setting up a voice over recording studio. It lists 8 steps to remember as a technical team, including soundproofing the room, managing noise levels, arranging your work station, and equipment needed. Key equipment includes a laptop, audio mixer, microphone, music stand, and monitor speakers. The document then describes how to set up the audio recording equipment, explaining the common controls on a mixing console like channel inputs, gain control, equalization controls, and master volume. It emphasizes never giving up and having confidence in your work.
This document provides information about technical stage operations related to sound equipment. It includes diagrams and explanations of how a basic PA system works, including sound sources, the sound desk, amplifiers, and active or passive speakers. It describes the inputs, outputs, and controls of a sample sound desk. It also covers topics like equalizers, reverb, delay, auxiliary controls, and different types of input cables.
This document contains information about various audio equipment including microphones, audio interfaces, studio monitors, headphones, and a MIDI keyboard controller. It provides specifications for the Line 6 TonePort UX1 and M-Audio Fast Track audio interfaces, Shure SM58 microphone, Behringer C-1 and C-2 condenser microphones, M-Audio Axiom 49 and KeyRig49 MIDI keyboard controllers, Sony earbuds, Sennheiser headphones, Xiaomi earphones, Samson MediaOne studio monitors, and Behringer Tube Ultragain Mic100 microphone preamplifier.
There are two types of speakers: active speakers, which contain an internal amplifier and only require a power source and sound input, and passive speakers, which do not contain an amplifier and require an external amplifier and connections to a sound source. Different cables are used to connect speakers to audio equipment, including DMX cables between speakers and sound desks, phono cables between sound sources and mixing desks, and various jack and power cables. A mixing desk controls audio by adjusting properties like volume, EQ levels, and panning for each input source and sending the processed audio signals to left and right speakers.
This document discusses frequency and its applications in equalization (EQ) and sound synthesis. It defines frequency as the number of cycles per second and describes how frequency is perceived as pitch. It discusses units of measurement like Hertz and how humans can typically hear frequencies between 20-20,000 Hz. It then covers topics like the frequency ranges of instruments, EQ, octaves, EQ bands, different types of EQ like parametric EQ, shelf EQ, and notch filters. It includes assignments to experiment with EQ in iTunes and record vocals to sound like they are coming from a telephone.
A compact, four channel field recorder capable of professional quality recordings. It supports up to 24-bit/192kHz sample rates and large capacity SDHC cards. Built-in effects, filters, and limiters allow for flexible audio editing and cleanup. Connectivity includes XLR, 1/4 inch, and digital inputs/outputs. Up to eight channels can be recorded by linking two units. The lightweight design runs on AA batteries or USB power, making it a portable solution for location sound.
RTASC Lite - Real Time Audio System Check LiteDru Wynings
This document provides instructions for performing a Real-Time Audio System Check (RTASC) using Audio Weaver Designer software to test the functionality of hardware and software for audio products with microphones and speakers. The RTASC is a step-by-step process that checks microphone matching, sensitivity and isolation, speaker distortion, microphone noise floor, conducted noise, and latency. It includes screenshots from the Audio Weaver Designer interface and templates for recording test results. Completing the RTASC can help identify issues and ensure the DSP Concepts voice frontend will perform properly.
The document provides an overview of an interactive voice conference on voice processing theory and algorithms for successful smart speakers and voice-enabled products. The agenda includes discussions on voice recognition algorithms, audio front-end processing, trigger word detection, beamforming, noise reduction, acoustic echo cancellation, and considerations for microphone and speaker integration in product design. Performance metrics and factors that affect various voice processing techniques are also outlined.
This document provides information on hardware and software requirements for Serato Scratch Live, as well as summaries of various Rane DJ interface and mixer products that are compatible with Scratch Live. It discusses the SL2, SL3, and SL4 interfaces, which allow connection of turntables and CDJs to a computer running Scratch Live. It also provides overviews of the Sixty-One, Sixty-Two, and Z-Trip Edition Sixty-Two mixers, which integrate with Scratch Live and provide controls for library, cues, loops and samples directly from the mixer. Specifications and features of each product are highlighted.
http://syntheway.com/Echo-Chamber-Reverb-Delay.htm - 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.
These VST and VST3 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).
- 12 Reverb Presets: Predefined space types, giving a virtually infinite number of possible shapes and sizes. These presets consist of some parameters to determine apparent room size, reverberation time, distance from you to the performer, etc.
- 4 Delay Presets: Doubling Echo, Slapback Echo, Ping-Pong Delay and Analog Tape Delay.
Preset Effects List:
01 • Room Reverb
02 • Hall Reverb
03 • Chamber Reverb
04 • Plate Reverb
05 • Spring Reverb
06 • Ambience Reverb
07 • Church Reverb
08 • Arena Reverb
09 • Opera Reverb
10 • Theater Reverb
11 • Cathedral Reverb
12 • Cave Reverb
13 • Doubling Echo
14 • Slapback Echo
15 • Ping-Pong Delay
16 • Analog Tape Delay
The S-Zone is a 4-channel mixer that allows mixing of audio signals from microphones, CD players, and more. It distributes independent mixes to 4 stereo zone outputs. Each input channel is assignable to any or all zones. The microphone inputs can lower ("duck") the background music volume during announcements. Equalizers and meters are provided on each zone output. Unique features include monitoring any zone through a front panel speaker and ability to remotely control zone volumes up to 3000 feet away.
The PLN-2AIO120 is an all-in-one background music and paging system that includes a DVD/CD player, FM/AM tuner, microphone inputs, and dual-zone amplifier. It provides versatile playback of audio and video content for background music and announcements. Key features include MP3/video playback from discs, an FM/AM radio, priority microphone input, and separate volume control for two speaker zones.
Chordophonet is designed to emulate the concert pedal harp, Celtic harp, electric and synth harp as well as an acoustic and electric hammered dulcimer. Includes a set of 20 pre-recorded harp glissando, two harp arpeggios, harp trill plus two hammered dulcimer glissando presets. Available as plugin in VST 32 bit and 64 bit and VST3 64 bit versions for Windows as well as in Audio Unit, VST and VST3 for macOS. Also available in EXS24 and KONTAKT Sample Libraries.
Sonifex Redbox are a range of audio and video interfaces designed for use in critical broadcast applications where the product needs to be powered 24 hours a day, 365 days a year.
DAL Flute & Woodwinds is an orchestral and ethnic woodwind virtual instrument collection, consisting of flute, oboe, clarinet, bassoon, piccolo, cor anglais (English horn), recorder, paixiao, dizi bangdi, shakuhachi, shinobue, quena, siku, nai, ney, ocarina as well as a small orchestra ensemble and woodwind section. Available in VST 32 bit and 64 bit and VST3 64 bit versions for Windows / Audio Unit, VST and VST3 for macOS. Also developed as EXS24 and KONTAKT Sample Libraries.
01 Flute Legato
02 Flute Non Vibrato
03 Flute Sustain Vibrato
04 Flute Staccato
05 Flute Staccatissimo
06 Flute Pizzicato (Slap Tongue)
07 Flute Trills
08 Oboe
09 Piccolo
10 Bassoon
11 Clarinet
12 Cor Anglais (English Horn)
13 Recorder (English Flute)
14 Shakuhachi (Japanese Flute)
15 Shinobue (Japanese Flute)
16 Paixiao (Chinese Panpipe)
17 Dizi Bangdi (Chinese Flute)
18 Andean Quena (South America)
19 Andean Siku (South America)
20 Siku Panpipe (Edge-Blown)
21 Bamboo Panpipe (Edge-Blown)
22 Nai (Romanian Pan Flute)
23 Ney (Ancient Persian Flute)
24 Ocarina (Sweet Potato)
25 Ocarina Vibrato
26 Orchestral Woodwinds (Flute, Oboe, Cor Anglais / French Horn, Bassoon and Contrabassoon)
27 Woodwind Section (Oboe, Cor Anglais / French Horn, Bassoon and Contrabassoon)
V.V.& Sons is the authorized distributor of RCF Loud Speakers and Sub Woofers in UAE and GCC Countries.
Contact Person - Joseph
Email ID - proinfo@vvsons.ae
Contact Number - 0097142684575
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.
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.
The Doro Secure 347 is a user-friendly telephone designed for easy and reliable communication. It features a remote answering function that allows calls to be answered remotely or for a preset alert number to be dialed. There is also a visual indicator light that flashes when calls come in. The volume can be adjusted on the handset, speaker, and ringer. It includes features like a speakerphone, photo memory buttons, and an assistance button.
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.
The document discusses audio consoles and their components and functions. An audio console combines and balances incoming audio signals and routes them to outputs. Key sections include inputs, outputs, monitors, and a mixer section. Analog consoles are described as having lower costs but easier operation than digital consoles, which provide features like automatic feedback suppression and noise resistance. The signal flow through a typical analog console channel strip is explained, covering components like the input section with trim knobs and pads, auxiliary sends, routing, channel faders, and equalization. Meter types like VU and peak are also defined. Finally, the Soundcraft Vi7000 digital console is briefly described.
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.
Sound is a multi-layered mixture of various elements. All things audio make up the soundtrack, and each has their own respective priorities, contexts, and purpose.
This document defines various types of diegetic and non-diegetic sound used in films. It discusses diegetic sound that exists within the scene's world and non-diegetic sound that does not. It also defines synchronous and asynchronous sound, parallel and contrapuntal sound, ambient sound, sound bridges, and sound motifs. Finally, it discusses the concept of verisimilitude, which is creating believability in a story through realistic details like sound. An example given is the sounds used in Saving Private Ryan to make the scenes as realistic as possible.
Dubbing involves mixing additional sound recordings with original production sound to create the finished soundtrack. During filming, diegetic sounds may be faded to highlight important character dialogue. This seems realistic even though it's not entirely accurate because the sounds establish atmosphere and involve the audience. A mixed soundtrack with adjustments to levels, equalization, panning and effects enhances scenes and storytelling. The dubbing process physically controls sound on a dub stage where mixers balance dialogue, effects, music and more to record the final track. Controlling sound aims to stimulate reality, create illusions and set moods that immerse audiences in the world of the film.
RTASC Lite - Real Time Audio System Check LiteDru Wynings
This document provides instructions for performing a Real-Time Audio System Check (RTASC) using Audio Weaver Designer software to test the functionality of hardware and software for audio products with microphones and speakers. The RTASC is a step-by-step process that checks microphone matching, sensitivity and isolation, speaker distortion, microphone noise floor, conducted noise, and latency. It includes screenshots from the Audio Weaver Designer interface and templates for recording test results. Completing the RTASC can help identify issues and ensure the DSP Concepts voice frontend will perform properly.
The document provides an overview of an interactive voice conference on voice processing theory and algorithms for successful smart speakers and voice-enabled products. The agenda includes discussions on voice recognition algorithms, audio front-end processing, trigger word detection, beamforming, noise reduction, acoustic echo cancellation, and considerations for microphone and speaker integration in product design. Performance metrics and factors that affect various voice processing techniques are also outlined.
This document provides information on hardware and software requirements for Serato Scratch Live, as well as summaries of various Rane DJ interface and mixer products that are compatible with Scratch Live. It discusses the SL2, SL3, and SL4 interfaces, which allow connection of turntables and CDJs to a computer running Scratch Live. It also provides overviews of the Sixty-One, Sixty-Two, and Z-Trip Edition Sixty-Two mixers, which integrate with Scratch Live and provide controls for library, cues, loops and samples directly from the mixer. Specifications and features of each product are highlighted.
http://syntheway.com/Echo-Chamber-Reverb-Delay.htm - 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.
These VST and VST3 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).
- 12 Reverb Presets: Predefined space types, giving a virtually infinite number of possible shapes and sizes. These presets consist of some parameters to determine apparent room size, reverberation time, distance from you to the performer, etc.
- 4 Delay Presets: Doubling Echo, Slapback Echo, Ping-Pong Delay and Analog Tape Delay.
Preset Effects List:
01 • Room Reverb
02 • Hall Reverb
03 • Chamber Reverb
04 • Plate Reverb
05 • Spring Reverb
06 • Ambience Reverb
07 • Church Reverb
08 • Arena Reverb
09 • Opera Reverb
10 • Theater Reverb
11 • Cathedral Reverb
12 • Cave Reverb
13 • Doubling Echo
14 • Slapback Echo
15 • Ping-Pong Delay
16 • Analog Tape Delay
The S-Zone is a 4-channel mixer that allows mixing of audio signals from microphones, CD players, and more. It distributes independent mixes to 4 stereo zone outputs. Each input channel is assignable to any or all zones. The microphone inputs can lower ("duck") the background music volume during announcements. Equalizers and meters are provided on each zone output. Unique features include monitoring any zone through a front panel speaker and ability to remotely control zone volumes up to 3000 feet away.
The PLN-2AIO120 is an all-in-one background music and paging system that includes a DVD/CD player, FM/AM tuner, microphone inputs, and dual-zone amplifier. It provides versatile playback of audio and video content for background music and announcements. Key features include MP3/video playback from discs, an FM/AM radio, priority microphone input, and separate volume control for two speaker zones.
Chordophonet is designed to emulate the concert pedal harp, Celtic harp, electric and synth harp as well as an acoustic and electric hammered dulcimer. Includes a set of 20 pre-recorded harp glissando, two harp arpeggios, harp trill plus two hammered dulcimer glissando presets. Available as plugin in VST 32 bit and 64 bit and VST3 64 bit versions for Windows as well as in Audio Unit, VST and VST3 for macOS. Also available in EXS24 and KONTAKT Sample Libraries.
Sonifex Redbox are a range of audio and video interfaces designed for use in critical broadcast applications where the product needs to be powered 24 hours a day, 365 days a year.
DAL Flute & Woodwinds is an orchestral and ethnic woodwind virtual instrument collection, consisting of flute, oboe, clarinet, bassoon, piccolo, cor anglais (English horn), recorder, paixiao, dizi bangdi, shakuhachi, shinobue, quena, siku, nai, ney, ocarina as well as a small orchestra ensemble and woodwind section. Available in VST 32 bit and 64 bit and VST3 64 bit versions for Windows / Audio Unit, VST and VST3 for macOS. Also developed as EXS24 and KONTAKT Sample Libraries.
01 Flute Legato
02 Flute Non Vibrato
03 Flute Sustain Vibrato
04 Flute Staccato
05 Flute Staccatissimo
06 Flute Pizzicato (Slap Tongue)
07 Flute Trills
08 Oboe
09 Piccolo
10 Bassoon
11 Clarinet
12 Cor Anglais (English Horn)
13 Recorder (English Flute)
14 Shakuhachi (Japanese Flute)
15 Shinobue (Japanese Flute)
16 Paixiao (Chinese Panpipe)
17 Dizi Bangdi (Chinese Flute)
18 Andean Quena (South America)
19 Andean Siku (South America)
20 Siku Panpipe (Edge-Blown)
21 Bamboo Panpipe (Edge-Blown)
22 Nai (Romanian Pan Flute)
23 Ney (Ancient Persian Flute)
24 Ocarina (Sweet Potato)
25 Ocarina Vibrato
26 Orchestral Woodwinds (Flute, Oboe, Cor Anglais / French Horn, Bassoon and Contrabassoon)
27 Woodwind Section (Oboe, Cor Anglais / French Horn, Bassoon and Contrabassoon)
V.V.& Sons is the authorized distributor of RCF Loud Speakers and Sub Woofers in UAE and GCC Countries.
Contact Person - Joseph
Email ID - proinfo@vvsons.ae
Contact Number - 0097142684575
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.
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.
The Doro Secure 347 is a user-friendly telephone designed for easy and reliable communication. It features a remote answering function that allows calls to be answered remotely or for a preset alert number to be dialed. There is also a visual indicator light that flashes when calls come in. The volume can be adjusted on the handset, speaker, and ringer. It includes features like a speakerphone, photo memory buttons, and an assistance button.
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.
The document discusses audio consoles and their components and functions. An audio console combines and balances incoming audio signals and routes them to outputs. Key sections include inputs, outputs, monitors, and a mixer section. Analog consoles are described as having lower costs but easier operation than digital consoles, which provide features like automatic feedback suppression and noise resistance. The signal flow through a typical analog console channel strip is explained, covering components like the input section with trim knobs and pads, auxiliary sends, routing, channel faders, and equalization. Meter types like VU and peak are also defined. Finally, the Soundcraft Vi7000 digital console is briefly described.
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.
Sound is a multi-layered mixture of various elements. All things audio make up the soundtrack, and each has their own respective priorities, contexts, and purpose.
This document defines various types of diegetic and non-diegetic sound used in films. It discusses diegetic sound that exists within the scene's world and non-diegetic sound that does not. It also defines synchronous and asynchronous sound, parallel and contrapuntal sound, ambient sound, sound bridges, and sound motifs. Finally, it discusses the concept of verisimilitude, which is creating believability in a story through realistic details like sound. An example given is the sounds used in Saving Private Ryan to make the scenes as realistic as possible.
Dubbing involves mixing additional sound recordings with original production sound to create the finished soundtrack. During filming, diegetic sounds may be faded to highlight important character dialogue. This seems realistic even though it's not entirely accurate because the sounds establish atmosphere and involve the audience. A mixed soundtrack with adjustments to levels, equalization, panning and effects enhances scenes and storytelling. The dubbing process physically controls sound on a dub stage where mixers balance dialogue, effects, music and more to record the final track. Controlling sound aims to stimulate reality, create illusions and set moods that immerse audiences in the world of the film.
This document contains a collection of 8 photos from Flickr shared under various Creative Commons licenses. The photos show a variety of subjects including food, machinery, landscapes, and more. Attribution is required for some while others have additional restrictions around commercial or derivative use.
1. The document provides advice on how to conquer the world or achieve anything you want by abandoning your hypotheses, expectations and comfort zone.
2. It suggests focusing on your goals and desires rather than what others think you should do. The only things you need to abandon are your assumptions, expectations and the zone of comfort that keeps you far from greatness.
3. By living according to your own rules rather than what is imposed on you, you can acquire the world or realize any other goal. You have the power to change your life and the world simply by doing what you want to do.
Donald Trump secured the Republican nomination for president after a contentious primary season against several establishment candidates. He campaigned as an outsider promising to "make America great again" through restricting immigration and renegotiating trade deals to benefit American workers. The election is shaping up to be one of the most consequential in modern history and will likely have broad implications for domestic and foreign policy.
Relationship marketing is a form of marketing that emphasizes customer retention and satisfaction over sales transactions. It recognizes the long-term value of customer relationships by using communication beyond just advertising to build more collaborative relationships. As technology has advanced, relationship marketing has evolved, using tools to gain insights beyond basic customer data to better manage customer relationships. Many types of companies can benefit from relationship marketing, including small businesses that rely on regular customers, large companies that invest heavily in relationship marketing campaigns, and industry leaders that must keep existing customers to maintain their position.
1) The Indian equity markets ended lower for the third straight session as key indices like the BSE Sensex and NSE Nifty declined 0.54% and 0.66% respectively due to disappointing industrial production data and weakness in index heavyweights.
2) Global markets also declined with the Dow and S&P 500 posting their sixth straight weekly losses on fears of a global economic slowdown and concerns about Greece's debt crisis.
3) On the corporate front, Bharti Enterprises announced plans to sell its insurance joint ventures to Reliance Industries while Tata Motors received an order worth Rs. 150 crore for trucks.
The document provides details about the planning, creation, and evaluation of a digital library project on musicals created by three students. It describes the planning process for selecting materials and intended users. Digitization and metadata processes are outlined, with ContentDM chosen as the platform. Pros of ContentDM included its user support and customization options, while problems encountered included technical compatibility issues. Lessons for future projects include appointing a group leader for coordination and clarifying roles and responsibilities.
The what, where and why of VR. Delivered at Augmented World Expo (AWE) in Santa Clara, this presentation highlighted the key do's and don'ts for VR Filmmaking, Marketing, Education and Retail. Also introducing the first #StereocastVR teaser reel.
360º #StereocastVR teaser available in Facebook 360 at: https://www.facebook.com/HelloBrandwidth/videos/10153483326630248/
More #StereocastVR info: https://hellobrandwidth.com/news/stereocastvr/
Client Training: Glassdoor's New Job Search ExperienceGlassdoor
This document summarizes Glassdoor's new job search experience. It includes an agenda for a webinar on the topic presented by Kelly Payne. The new experience features a more intuitive job search interface that allows users to view more job descriptions. All jobs will now be hosted on Glassdoor, including employer ratings, reviews and salary information. Sponsored jobs can include branded employer content within the job description. Key benefits are attracting informed candidates, increasing employer brand awareness, and helping employers stand out from competitors.
Osama Mohamed Ahmed Bader is seeking a challenging position in a strong, growing company where he can utilize his skills and training. He has a Bachelor's degree in Accounting and work experience as a call center agent and accountant. His training includes courses in financial analysis, ICDL, English, human development, and computer skills. He has strong communication skills, can work well in teams, and is a hard worker with high flexibility.
Sound is a longitudinal wave that travels through a medium as a series of compressions and rarefactions. When sound waves reach the ear, they are converted into electrical signals that travel to the brain via auditory nerves. Microphones convert sound waves into electrical signals. Microphone placement depends on the sound source and microphone characteristics to capture sound with proper clarity, balance, and lack of feedback. The document discusses the nature of sound waves, microphone types and their uses, and considerations for microphone placement.
Report on bass and trable - Analog ProectVatsal N Shah
This project report describes designing a bass/treble separator circuit to separate low and high frequency audio signals and output them to separate lines for a woofer and tweeter. The circuit uses a low-pass filter with 150Ω resistor and 10μF capacitor to output bass frequencies to the woofer, and a high-pass filter with the same components to output treble frequencies to the tweeter. The project was carried out by a student at Indus University for their 4th semester electronics course.
This document discusses sampling theory and digitizing sound. It explains:
- How sound can be represented in the time and frequency domains.
- The Nyquist-Shannon sampling theorem, which states that a signal must be sampled at least twice the highest frequency to avoid aliasing.
- Key parameters for digitizing sound like sampling frequency, bit depth, and their effects on quality and file size.
- Common digital audio standards and transmission speeds like CD, telephone, ISDN, T1, and how they relate to sampling theory.
Sound level meters are instruments used to objectively measure sound pressure levels. They respond to sound similarly to the human ear and provide reproducible measurements in decibels. Sound level meters comprise a microphone that converts sound to electrical signals, along with signal processing circuits that apply time and frequency weightings to simulate human hearing. Measurements can be taken using different time weightings like fast, slow, and impulse. Frequency weightings like A, C, and Z are also used to adjust for the ear's sensitivity. Sound level meters are applied to measure noise in environments like workplaces and construction sites.
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 various topics related to architectural acoustics including:
- The definition of architectural acoustics as the study of sound generation, propagation, and transmission in buildings.
- The importance of applying acoustic principles to improve quality of life through work and leisure environments.
- The need to both enhance desirable sounds like music, while reducing undesirable noise.
Sampling rate refers to the number of digital samples taken per second of an analog audio signal. A higher sampling rate allows for more accurate reproduction of the original sound by capturing more data. The standard CD sampling rate is 44.1kHz.
Bit depth determines the number of possible amplitude levels that can be represented in each digital sample. A higher bit depth provides more precision in capturing the amplitude but requires more storage space. Standard CD audio has a bit depth of 16-bits, providing 65,536 possible amplitude levels per sample.
When an analog audio signal is converted to digital, the continuous waveform is converted into discrete samples. The difference between the original analog signal and the quantized digital representation is called quantization error
This document discusses key concepts in digital audio, including:
1) Digital audio is discrete in both time and amplitude, where analog is continuous. Sampling converts an analog signal to digital by taking discrete time and amplitude samples.
2) For lossless sampling, the sampling rate must be at least twice the bandwidth of the analog signal to avoid aliasing.
3) Quantization converts the sampled amplitude values to discrete digital values. More bits provide higher resolution and dynamic range but introduce quantization error and noise.
4) Digital audio can be transmitted via various standards like AES/EBU and S/PDIF using pulse code modulation to encode the digital samples into a binary data stream. O
PURE TONE AUDIOMETRY for assesment of hearingVaishnawiRai
Pure tone audiometry is used to evaluate hearing thresholds and detect hearing loss. It involves presenting pure tones of varying frequency and intensity to determine the softest level a person can hear at each frequency. The results are plotted on an audiogram. Key components of pure tone audiometry include calibrated equipment to generate tones, different transducers for air and bone conduction testing, and masking noise to isolate each ear. Interpretation of the audiogram can identify normal hearing, conductive hearing loss, sensorineural hearing loss, or mixed hearing loss based on the air and bone conduction thresholds and air-bone gap. Different audiogram patterns provide clues to the possible cause of any detected hearing loss.
FUNDAMENTAL ACOUSTICS AND WIND TURBINE NOISE ISSUESriseagrant
Wind turbines produce noise from aerodynamic and mechanical sources. Aerodynamic noise from airflow over the blades is the largest contributor. The sound is amplitude modulated by blade rotation. Wind turbine noise is perceived as more annoying than constant noise due to its unpredictable nature. Noise levels decrease with distance from the turbine following laws of spherical spreading and atmospheric absorption. Low frequency noise and infrasound may be issues for some turbines operating downwind of towers. Regulations establish noise limits and setback distances to minimize community impact.
FUNDAMENTAL ACOUSTICS AND WIND TURBINE NOISE ISSUESriseagrant
Wind turbines produce noise from aerodynamic and mechanical sources. Aerodynamic noise from airflow over the blades is the main contributor. The noise is amplitude modulated by blade rotation and varies with wind speed. Wind turbines can be perceived as intrusive in rural areas with low background noise. Noise regulations set limits on allowable sound levels and require setbacks from residences. Low frequency noise and infrasound may be issues for some downwind turbine designs.
A pure tone audiometry test is used to find out actual hearing levels as well as type and degree of hearing loss by means of two pathways the Air conduction and Bone conduction.
This document discusses front-end audio processing techniques used in communication and recording devices. It provides an overview of classical front-end architectures used in phones from the 1990s to 2010, including filters, equalizers, encoders/decoders, volume control, acoustic echo cancellation, noise suppression, and leveling. It also discusses issues like dynamic range, quantization noise, distortion, and the tradeoffs between noise suppression and voice quality. The document reviews single-channel noise suppression techniques and metrics like convergence rate and PESQ/MOS-LQO. Finally, it introduces multi-microphone solutions like linear beamforming and differential beamforming and their benefits and limitations regarding critical distance and noise.
This document discusses key concepts related to sound waves and digital audio. It covers topics like the parts of a sound wave, pitch and frequency, the audio range, musical pitch, wavelength, octaves, the frequency spectrum, phase, timbre, sound envelopes, acoustics, direct vs indirect sound, studio design, room acoustics, analog vs digital recordings, sampling, bit depth, quantization, AD/DA conversion, digital audio workstations, audio formats, uncompressed vs compressed files, and sample rates.
Sampling rate refers to the number of times per second that the amplitude of a sound wave is recorded in a digital audio file. A higher sampling rate allows for higher frequencies to be captured, more closely representing the original sound. The standard CD sampling rate is 44.1 kHz. Bit depth refers to the number of possible values used to record the amplitude of each sample, with 16 bits being standard for CD quality audio. Higher bit depths can capture a wider dynamic range of sounds but take up more storage space. Lossy compression formats like MP3 and Vorbis reduce file sizes by removing some audio information considered imperceptible to human hearing.
This document discusses the basics of digital audio, including analog vs. digital representations of sound, sampling, sampling rate, Nyquist frequency, aliasing, quantization, dynamic range, and quantization error. It explains that analog audio is continuous in both time and amplitude, while digital audio represents sound in discrete time intervals and amplitude levels through the processes of sampling, quantization, and reconstruction. Key terms like sampling rate, Nyquist frequency, aliasing, word length, dynamic range, and quantization error are defined.
The document discusses the analog to digital conversion process. It explains that sounds are analog waves but computers are digital so an conversion is needed. The sound card contains an analog-to-digital converter (ADC) that samples sounds and converts them to binary digits and a digital-to-analog converter (DAC) that converts the digital signals back to analog waves for playback. The key parameters for conversion are the sampling rate, which must be over twice the highest frequency to avoid quality issues, and the bit depth, which determines the number of possible values and thus the resolution/quality. Higher rates and depths allow for better quality recordings.
Sound waves are vibrational disturbances that transmit energy through a medium by compressing and rarifying the molecules of that medium. The frequency of a sound wave is measured in Hertz (Hz) and determines its pitch, with human hearing ranging from 20-20,000 Hz. Below 20 Hz are infrasonic sounds like some animal calls, and above 20,000 Hz are ultrasonic sounds like dog whistles. The fundamental frequency is the lowest (or basic) frequency that a sound source can produce.
This document discusses key concepts in capturing and working with audio digitally. It explains how sound is converted from acoustic waves to electrical voltages through microphones, then from analog voltages to digital numerical values using analog-to-digital converters. It describes common sample rates used in audio recording and formats like CDs, DVDs, how higher sample rates provide more accuracy. It also covers bit-depth and how it determines the dynamic range, and how sample rate and bit-depth together affect audio quality and file size, like with digital camera resolution. Finally, it mentions visualizing audio as waveforms and common file formats like WAV, MP3, and AAC.
Advice to students about to graduate and looking to get started in the film industry. Guide to creating your resume, how to network and get noticed, making the most of trade shows and film festivals, creating your own internship, and how to survive working deferred on your first films.
A brief presentation outlining the basic operation and use of the Zoom H4n digital 2-track recorder. A very popular unit well suited for double system sound recording for film and video applications.
The document provides instructions for the process of calling the roll at the beginning of a film or video shoot. It describes the steps taken by the assistant director, sound mixer, camera operator, and clapper/loader. These include putting the set on a light and bell, rolling sound or video, slating the scene and take number, recording room tone, marking the clapstick, and calling action. The goal is to synchronize footage and audio while providing scene identification for editing.
This document introduces the importance of sound in movies from early silent films to modern productions. It discusses the transition to "talking pictures" and how dialogue is now a key part of film scripts and storytelling, as the soundtrack alone can convey much of the plot without video. Production sound is crucial for recording live dialogue on set.
There are three main classes of microphones: dynamic, electret condenser, and true condenser. Dynamics have the least sensitivity and generate electricity from sound-induced movement within a magnetic field. Electret condensers require battery power and have greater sensitivity than dynamics. True condensers are the most sensitive and require external phantom power but are best for capturing subtle sounds. Microphone selection depends on the application and a balance of attributes like reach, sensitivity, pattern, and rejection of ambient noise and echoes.
There are several microphone patterns that determine the directionality of sound pickup. Omni-directional microphones pick up sound equally from all directions, making them good for interviews where mic angle is variable. Cardioid microphones are most sensitive to sound from the front and pick up less from the sides and back, reducing feedback and ambient noise. Bi-directional microphones pick up sound equally from the two sides but not the front or back. Mid-side stereo techniques use a bi-directional mic together with a cardioid mic to allow control over the left-right stereo image during mixing. Hypercardioid and shotgun microphones have a tight, directional pickup pattern that focuses on sound from directly in front
The document discusses different types of microphones and their common applications. It describes microphones used for podiums and desktop recording that are dynamic and prevent feedback. It also mentions lavalier microphones that are small and body-worn, as well as pressure zone microphones that are mounted above a hard surface to pick up sound hemispherically from several feet away. Studio music microphones are highlighted as high quality condenser microphones suitable for rich sound recording but too sensitive for live performance.
The document outlines six priorities for a production mixer when recording audio on set: 1) Getting usable dialogue is the top priority. 2) Maintaining proper audio perspective to match the camera viewpoint. 3) Syncing sound effects without compromising dialogue. 4) Recording wild lines and off-camera dialogue for coverage. 5) Capturing wild sound effects not available in libraries. 6) Recording ambiance and room tone to patch holes in the edited soundtrack.
This document provides guidance for sound crew roles and responsibilities on film productions. It discusses the roles of the production mixer, boom operator, and utility sound technician. It covers salaries, equipment needs, budgets, and best practices for professional and effective sound capture and management. The key roles, responsibilities, and priorities are to hire the right crew, properly prepare equipment, capture all dialogue, and maintain a professional attitude and efficient workflow.
Production sound is the complex craft of recording live dialogue and sound effects on set during filming. It requires a blend of creative judgement and technical expertise to deal with challenges like noise problems, equipment failures, and changes in blocking or performance. The production sound mixer fills several roles, operating audio equipment while also serving as a sound editor, director, and member of the filmmaking team.
Production sound is an essential element of filmmaking that provides more employment opportunities and a higher position in the crew hierarchy than other jobs like camera, editing, or production assistant. Learning production sound helps directors understand crew capabilities to effectively communicate priorities and evaluate audio quality, and it helps those in the film industry understand sound crew roles to work as a successful team.
Exploiting Artificial Intelligence for Empowering Researchers and Faculty, In...Dr. Vinod Kumar Kanvaria
Exploiting Artificial Intelligence for Empowering Researchers and Faculty,
International FDP on Fundamentals of Research in Social Sciences
at Integral University, Lucknow, 06.06.2024
By Dr. Vinod Kumar Kanvaria
This slide is special for master students (MIBS & MIFB) in UUM. Also useful for readers who are interested in the topic of contemporary Islamic banking.
How to Setup Warehouse & Location in Odoo 17 InventoryCeline George
In this slide, we'll explore how to set up warehouses and locations in Odoo 17 Inventory. This will help us manage our stock effectively, track inventory levels, and streamline warehouse operations.
Walmart Business+ and Spark Good for Nonprofits.pdfTechSoup
"Learn about all the ways Walmart supports nonprofit organizations.
You will hear from Liz Willett, the Head of Nonprofits, and hear about what Walmart is doing to help nonprofits, including Walmart Business and Spark Good. Walmart Business+ is a new offer for nonprofits that offers discounts and also streamlines nonprofits order and expense tracking, saving time and money.
The webinar may also give some examples on how nonprofits can best leverage Walmart Business+.
The event will cover the following::
Walmart Business + (https://business.walmart.com/plus) is a new shopping experience for nonprofits, schools, and local business customers that connects an exclusive online shopping experience to stores. Benefits include free delivery and shipping, a 'Spend Analytics” feature, special discounts, deals and tax-exempt shopping.
Special TechSoup offer for a free 180 days membership, and up to $150 in discounts on eligible orders.
Spark Good (walmart.com/sparkgood) is a charitable platform that enables nonprofits to receive donations directly from customers and associates.
Answers about how you can do more with Walmart!"
This presentation was provided by Steph Pollock of The American Psychological Association’s Journals Program, and Damita Snow, of The American Society of Civil Engineers (ASCE), for the initial session of NISO's 2024 Training Series "DEIA in the Scholarly Landscape." Session One: 'Setting Expectations: a DEIA Primer,' was held June 6, 2024.
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This presentation includes basic of PCOS their pathology and treatment and also Ayurveda correlation of PCOS and Ayurvedic line of treatment mentioned in classics.
How to Build a Module in Odoo 17 Using the Scaffold MethodCeline George
Odoo provides an option for creating a module by using a single line command. By using this command the user can make a whole structure of a module. It is very easy for a beginner to make a module. There is no need to make each file manually. This slide will show how to create a module using the scaffold method.
The simplified electron and muon model, Oscillating Spacetime: The Foundation...RitikBhardwaj56
Discover the Simplified Electron and Muon Model: A New Wave-Based Approach to Understanding Particles delves into a groundbreaking theory that presents electrons and muons as rotating soliton waves within oscillating spacetime. Geared towards students, researchers, and science buffs, this book breaks down complex ideas into simple explanations. It covers topics such as electron waves, temporal dynamics, and the implications of this model on particle physics. With clear illustrations and easy-to-follow explanations, readers will gain a new outlook on the universe's fundamental nature.
2. • Vertical axis = AMPLITUDE (aka loudness)
• Horizontal axis = FREQUENCY
• Positive and Negative components of the wave
• Expressed as Hz or kHz. Example: 22 kHz (or 22k for
short)
3. • Digital audio is a graphic “snapshot of the sine wave”
• The “pixel grid” is divided into “vertical resolution” and
“horizontal resolution”.
• Vertical res are BITS, as in 16 bit, 24 bit, 32 bit floating.
• BITS actually refer to the size/detail of the data packet
• Horizontal res is SAMPLING RATE, slices per second, as
in 44.1 kHz, 48 kHz
4. • Sampling rates commonly range from 8 to 192.
• Low fidelity rates: 8, 11, 16, 22, 32. Poor freq response.
• Used for radio comms, telephone, cheap recorders,
along with lower bit rates.
• Higher sampling rates: 44.1, 48, 88.2, 96, 192
• 44.1 = consumer audio CD (.CDA)
• 48 = professional audio files, digital cinema/DCP
• 96 = uber quality (music) recording intended for future
editing and down conversion to 48 or 44.1
• 192 = NASA grade instrumentation recording of vibration
5. • “In order to get a clear “snapshot” of the freq response,
we need to sample it twice, and then subtract a portion
for housekeeping.
• “Twice, has something to do with the fact that each sound
wave has two parts to it, a positive curve and a negative
curve.
• “Housekeeping requires 2k.”
• Sampling rate, divide by two, subtract another 2k = Freq
Response. (Sample/2) minus 2 = Freq.
• *note: Prof Ginsburg’s words, not the actual
mathematical (geek speak) theorem.
6. • Example. 48k sampling rate
• 48/2 = 24
• 24 minus 2 = 22
• Freq response of 48k is only 22 kHz, as in 20-22 kHz
• Example. 44.1 sampling rate
• 44/2 = 22
• 22 minus 2 = 20
• Freq response of 44.1 k is only 20 kHz, as in 40-20 kHz
7. • Noise purposely added to the digital track to mask the
sterile, “on/off” listening experience of pure digital (1-0-1-
0) sampling.
• Kind of a way of smoothing off the rough edges, so to
speak.
• Think of it as adding a touch of diffusion to make a high
resolution “facial portrait” more flattering!
8. • Freq response approx
20-22kHz, but reduces as
we get older to just 12 k or
14 k.
• Pain threshold approx 120
dB spl (sound pressure
level)
• Hearing damage from 90
dB
• Avoid loud earphones,
monitor speakers!!!
• Hearing damage is
permanent.
9.
10. • A way to measure the volume (gain) of an audio signal
• Doubling the sound pressure (voltage) corresponds to a
measured level change of 6 dB. Doubling of sound
intensity (acoustic energy) belongs to a calculated level
change of 3 dB.
• Think of 3 dB as one F-stop of sound
• We perceive equal (voltage) levels of sound as different
“loudness” based on their frequencies, referred to as
Fletcher-Munson curves.
• We use weighted meters (such as A-weighted) to
compensate.
• Engineers use variations of the dB term to refer to actual
voltage levels of the audio signal. Too geeky to get into.
11. • 20 Hz to120 Hz = Bass.
• Most mics roll off around 80 Hz to filter out low freq noise such as
handling, wind, rumble/vibration, distant traffic, ventilation
systems.
• 100 Hz to 5 kHz = Mid-range. average range of human
voice
• 5 kHz to 6 kHz = Upper mid-range to lower High freqs.
• Sometimes we roll off sibilance (de-Essing) of human voice
• 6 kHz to 22 kHz = High frequencies. Harmonics.
12. • Reverb is characterized as random, blended repetitions
of a sound occurring within thirty milliseconds after the
sound is made. This is all the sound that immediately
bounces off any nearby surfaces before it gets back to
your ears.
• Echo is defined by distinct repetitions of a sound
occurring after 30 milliseconds. This is when you can
unquestionably hear a distinct... well, echo of a sound
coming back to you.
• Echo often is a diminished quality of the signal, due to
greater disproportional frequency loss and delay.
13. • Measurement of resistance. Means different things
depending on the context.
• Mic Level output, 250 Ω, a low level output.
• Line Level output, 600 Ω, a much louder output.
• Mic Level input, 250 Ω, expects a low level input.
• Line Level input, 600 Ω, wants a much louder input.
• Approx 50 dB difference in volume between mic v line
level
• Headphones, 40 to 100 Ω is ideal. Less than 35 > too
loud and easily distorted. More than 100 < provides not
enough volume from field recorders.
14. • Mic out to Mic in. Sounds OK, but weaker signal more
prone to interference.
• Line out to Line in. Sounds OK, a stronger signal less
prone to interference. Best settings to use if you can.
• Mic out to Line in. Very low and feint volume. Think of
sending 12v into a 100v light bulb.
• Line out to Mic in. Too powerful a signal for that input.
Audio will be loud and distorted. Think of 100v going into
a 12v light bulb.
• Line level is normal level for most devices. Mic level
requires pre-amp to raise level up to Line level for mixing
or recording.
15. • Why actors tell each other to “break a leg”
• Has nothing to do with fracturing one’s bones!
• During Medieval era, theater actors only got paid if they
appeared on stage that day.
• To “break a leg” meant that your leg cleared (aka broke)
the edge of the stage curtains and you would be visible in
the performance…
• So you were entitled to wages.