This is the mechanism that enables one’s depth perception. This is what a modern 3D camera seeks to emulate. These cameras use two lenses which allow a three-dimensional signal to be recorded by determining the differences between the two images which is analogous to how your brain decodes your own three-dimensional vision.  But these images are displayed on two-dimensional screens.  In order to give depth to the image, glasses are used.

It is possible that stabilizing the cameras with gyroscopically controlled panheads will alleviate some of the discomfiture issues, but that still leaves the rest of the sensory inputs to be concerned with. To my knowledge, there have been no definitive studies published on the effect of surround audio in combination with 3D imagery in terms of image stability. There have certainly been papers presented regarding localization of sound in a surround matrix, especially in respect to video game production. Studies indicate that audio information presented in 3D improves reaction time, as noted in certain high stress environments such as airplane cockpits.[1]  Will accurate 3D audio reproduction have a positive effect on viewer reaction to 3D disorientation?  It has yet to be seen.
3D technology is still relatively new, and is being studied and improved upon constantly.  Perhaps the answer may be to place the viewer in the center of a holographic matrix where one could possibly have a better connection between motor control and perception. 
Include Footnote Link (PDF) - Psyko Audio Labs , Surround Technologies for headphones.
I more or less cited this article when talking about the correlation between 3D audio and reaction time.  I have noticed that when mixing for TV - by having the various "instructions" (director/producer, etc) come from different speakers in different locations, I can react faster and understand better.
Daniel Littwin , director New York Digital.
contact: daniel.nydigital@gmail.com
São Paulo, Brazil

Mentioned:
NAB convention in Las Vegas

Tags: 3D

Technical Details:
The show was recorded in HD, 1080i and surround by All Mobile Video’s Crossroads OB unit for editing and transmission the following week in Italy. Richard Wirth directed the cameras. He and Lenny Laxer of AMV managed the technical aspects of the production. Twelve Sony HD1000 or HD1500 cameras were used, including two jibs, two fixed cameras, a remote controlled track-mounted robotic camera on a crane, a hand-held camera and a steadicam. The backdrop was an enormous (25m x 31m) HD LED screen that was provided by Radio City. This screen was fed with live images from the cameras as well as prerecorded scenes and images, all sequenced and timed to the show and controlled by a computer. The “line cut” and all of the cameras and images were individually recorded. Over one hundred seventy two audio channels were recorded, including the microphones used to capture the ambience and microphones mounted on the moving cameras. All of these microphones were mixed live as well as recorded individually in Protools for remixing in postproduction. Microphones and effect groups were created and organized by instrument type which were then sent to the Studer Vista 8 mixing desk in the OB UNIT. Certain key performance mics, like the lead vocal and certain soloists were sent individually. All audio routing in the OB unit originated in the Studer Vista 8.

Because of time and budget constraints we never had a full dress rehearsal. Richard, our director used a series of coverage zones for the cameras, similar to a multi-camera film shoot, so that each camera had a specific area to concentrate on. One camera covered the star, Gigi at all times. Other cameras covered the dance routines and guest artists. This ensured that all aspects of the performance could be video taped for use in post-production, if necessary. Stage entrances were pre-planned so that each guest could be shown entering and leaving the stage. Occasionally, these were altered at the last minute which caused some consternation in the OB unit. However, clear communication and fast thinking always saved the shots.

Intercommunications:
The intercommunications system was an RTS/Telex Adams digital matrix system with 144 inputs and outputs. In the theater, 12 channels of wireless intercommunication were used in addition to 26 wired stations and all the camera intercoms. All key personnel in the OB unit had 32 channel digital key panels.


Setting Up
The show was scheduled to start at 19:30 on Monday, 14 Feb. Set up started two days earlier at 08:00. We were allotted 10 hours on Saturday to install all of the equipment. On Sunday we had another 10 hours for rehearsals of specific camera angles for dance routines and specific timing issues as well as some fine-tuning of the equipment.

The main lighting array was installed first. The PA system was installed next. It was an i-Series line array, provided by Clair. The house audio desk was a Protools Venue. The monitor audio desks were Digico SD8’s. Protools HD was used to record the show. Word clock was derived from the video master clock on the mobile unit, which also provided time code. As the show was for broadcast in Italy, all of the video and time code references were PAL, based on 25 frames/second. The audio tracks were recorded at 48K/24 bit.

The client provided us with a wish list, which we translated (sometimes literally from Italian into English) into our technical requirements. Intercommunications systems are always essential: often overlooked, until there is a problem, but if nobody can communicate, the show will NOT “go on”. In this case, someone would usually say or write, for example, that “Roberto” needed to speak with “Julio”, “Annette” and “Richard”. We would have to determine who these people were, what they did and what language they spoke. This in and of itself was a challenge as there was no complete crew list of both technical and production people and nobody wore nametags. We would then create the necessary channel in the matrix software and label it accordingly. Modern intercommunications systems are both software and hardware based, so not only does the computer need to be programmed correctly, but also the hardware needs to be adjusted and balanced for optimum performance. In a live event, there is only one chance to capture the moment, so correct equipment preparation is of the utmost importance.

The sound reinforcement crew and the artist determined all the microphone selections and positions. All wireless frequency allocation was determined on site by using “Shure Wireless Workbench™” software. The microphones, IFB’s, and wireless intercoms were coordinated to provide the most reliable performance without radio frequency interference. As there were 38 wireless mics, 24 wireless monitors and 12 channels of wireless intercom, as well as literally hundreds of walkie-talkies, this coordination was critical to the success of the show.

The intercommunications system had to accommodate both English and Italian speakers, sometimes sharing the same channels. Stage managers needed to speak with the artists, the Italian production staff and the American technical staff.
The television and camera crew all spoke English, as did the director. However, the producer, lighting designer, choreographer and musical director were all Italian. A great deal of patience was needed (as well as hand gestures). One interesting challenge was to keep the critical technical conversations in English separate from the creative/production conversations in Italian. All departments were assigned their own channels for communication; lighting, PA, TV audio, house coordination, video, camera, engineering and two channels for production.

A note regarding labels and identification: as the technical staff was mostly American and the production staff was mostly Italian all of the equipment needed to be labeled in both languages. This was especially critical for communications. Only a very few people were fluent in both English and Italian.

Video Tape
All cameras and images were recorded individually to videotape. All machines were fed with timecode referenced to time of day. Every machine received a stereo mix of the program. All of the machines recording moveable cameras were fed with the particular camera’s associated microfone, copied to two channels. Certain image sequences were also recorded to EVS video hard disk systems in order to create playlists that were later recorded to VTR’s. Additionally, several HD camcorders were used to capture ambience from backstage, the audience and on the street.

Videotape ran continuously. All recordings were started at different times, so that overlaps could be recorded in EVS hard disk recorders and then transferred to videotape with the original timecode. All tapes were 1 hour in length. With a total of 16 recorders it would be impossible to change all tapes at once and keep a continuous recording of every source. So this system of overlaps was implemented. Using longer videotapes is not generally advisable as longer tapes are thinner and more subject to damage.

Multitrack Audio
Every input source on stage was captured separately and recorded to a Protools HD system. Most of the signals were derived from the FOH audio desk. Ambience and audience microfones were recorded directly to the Protools HD system, using separate microfone preamps.

Here is a list of all of the microfones used in recording:

The antennas, receivers and transmitters for the wireless microphones and in-ears were located next to the monitor audio desk, backstage left. Most of the interface between the PA system and the OB unit was located here as well. To ensure better RF separation we placed the wireless intercom on the opposite side of the backstage area. This allowed for greater flexibility in focusing the antennas for better coverage and better access to the equipment for programming and maintenance.
Every frequency of every channel had to be changed to fit into the available spectrum.
Very often, on larger shows frequency coordination is handled in advance: every major venue in the USA has been “mapped” to show what spectra are available. It is therefore merely a matter of cross checking the available frequencies against what your equipment is capable of. All that is required is a list of the available gear.
Intermodulation and illegal frequency and power use is fairly common, so an RF spectrum analyzer is used to confirm the available space. The equipment is then fine adjusted on site. In the case of this show, all frequency coordination and adjustment was done on site, which took a considerable amount of time.

Radio City Music Hall is a “union” house. The entire technical crew are members of
IATSE, the International Association of Stage and Theatrical Employees. The rules are fairly rigid, but the crew at Radio City are extremely competent, helpful and generally friendly. It can be intimidating to work at such a famous facility, but with patience and respect to the regulations of the theatre, Radio City is one of the best places in the world to work. The attention to detail and personal safety is both impressive and reassuring. For example, when the lighting grid is about to be lowered, a horn sounds and ALL work stops. This reduces confusion, allows the appropriate personnel to work quickly and efficiently and prevents injuries due to falling objects or heavy cables.

Radio City is renowned for its “stagecraft” Every part of the stage is on a hydraulic lift, so that entire sets can appear from and disappear into the floor, as if by magic.
Large objects, even trucks and elephants can be rolled, lifted into the ceiling, or flown across the stage. Of course our show did not include any flying elephants, but they would have been happily accommodated for, had they been needed for the production.
However, the capabilities of the house meant that we were able to build a very large show in about 16 hours, with an orchestra, band, dancers, special guests, complicated lighting and scenery cues, a large PA system, all of which had to be integrated with the High Definition OB unit with the Grass Valley Kalypso, Studer Vista 8, intercom system, and HD television cameras. Disassembling the show was also made easier by having such capabilities.

All of the necessary configurations were saved in all of the various mixing desks and recalled as each different pieces of music required. Lighting cues and video screen presentations were also preset. Camera shots were mostly handled manually, with a few presets on particular cameras for finding focus when the lights were out. Video settings for each camera were preset and then adjusted manually as required by particular shots. Cameras had to be maintained in correct adjustment constantly, as each camera was being recorded individually at all times.


The Show
The show was presented as one continuous performance without an intermission. Italian comedian Enrico Brignano provided some routines between musical segments. With all the performers and comedic intervals the entire show lasted about three hours, including the encores. The show was recorded with extensive notes and then edited in Italy for presentation the following week. The presentation was also to be made available for DVD release in Europe.

Daniel Littwin , director New York Digital.
contact: daniel.nydigital@gmail.com
São Paulo, Brazil

Tags: Audio, Gala Show

This is my first blog entry. So before I get to my subject, I’d ike to introduce myself. I am Danny Littwin, an American television tech manager, audio engineer and musician residing in São Paulo, Brazil. Before moving here in 2007, I was a US television freelancer, work that I still continue today occasionally. Most of my career has been as an audio engineer. I have mixed just about every kind of event imaginable, from live televised sports events to large orchestral presentations, from ice hockey to pop music and from football to traditional Irish music. For my efforts and through a combination of good timing and good luck I have received several awards: 6 Emmys, 3 Grammys and a bunch of nominations.


I think it is not so much the awards that count, but rather the fact that I have been able to help my clients and colleagues attain their artistic and commercial visions through my experience and intuition. TV is a great business to be in, as most of the people you work with are there because they enjoy it. You could certainly make more money as commodities trader, but then what fun is that? Some of my favorite memories are getting calls from my grandmother, telling me she saw my name in the credits.

Before I was a tech manager and director of a broadcast services company, I was an audio engineer. So today’s entry will deal with an interesting audio topic:

Maintaining Intelligibility in Principle Mix Sources

Tags: Audio

1. Microphone position
2. Choice of microphone
3. Isolation mounts for microphones
4. Gain and balance throughout the signal path
5. Selection of specific equipment:
1. Solid state or tube mic preamps, transformer, transformerless, etc.
6. Use of signal processing:
1. Equalization, dynamics, effects devices, etc.
7. Pan position and mix balance.

1. How many sources are in the mix?
1. How many of the sources are microphones?
2. Do all the microphones share the same space/time perspective?
3. Is this a live event or a multitrack/multilevel recording?
4. Is there a substantial amount of background noise?
5. How reverberant is the environment?
6. Are there competing points of interest?
7. Will this be reassembled in post-production or broadcast live or both?
8. Are there multiple mixing desks in operation? i.e.: transmission, recording and PA
2. What is the final destination of the product?
1. Radio or TV Broadcast?
1. Mono, stereo or surround?
2. What are the parameters of the metadata at the transmitter?
3. What are the parameters of the limiters at the transmitter?
2. CD, DVD or Video Game?
1. Is metadata being recorded?
2. What parameters are specified?
3. Internet streaming or MP3?
1. What bit rate?
3. What is the record media?
1. Hard drive, solid state or tape-based digital media?
2. Analog tape
1. With or without noise reduction?
2. What tape speed is being used?

Digital Versus Analog Distortion
Analog distortion can actually be used as a desirable effect. Analog distortion bears an inverse resemblance to the original audio signal, in that as more distortion is added, the signal waveform becomes less recognizable. Digital distortion is undesirable, as it bears no resemblance to the original signal. Furthermore, digital distortion is absolute above a given threshold and non-existent below that threshold. This also means that in response to a sudden transient, a signal can be distorted and unusable once the dynamic range of the system is surpassed.

Tape saturation, valve saturation and output clipping are all types of compression, which can be quite pleasant if used carefully. Transformers can have an effect similar to equalization, slightly emphasizing or de-emphasizing certain frequencies, depending on many factors including the core density, number of windings and wire gauge used in the transformer. There are also many digital plug-ins that emulate analog processors.
They can all be effective and complimentary if used within the dynamic range of the system.

In any mix, every signal affects every other signal. Good mixes take advantage of this phenomenon to create a new complimentary sound out of the elements presented. Distorted waveforms will combine with non-distorted signals and make the individual components of the mix appear to be corrupted. Distortion will affect not only frequency response but also signal phase as well. Therefore distortion will lead to irregularities or even a collapse of the stereo or surround image, which is based on phase information.

Data Compression, Noise Reduction and Signal Encoding/Decoding
If data compression is used it is imperative that the correct codecs be implemented. Most often this is handled automatically. In some cases it is necessary to choose. If noise reduction is used it must be consistent throughout the reproduction chain. When mixing in surround it is extremely important to manage the metadata correctly, both while encoding and decoding.

Accurate documentation is very important in maintaining signal quality. Regardless of how the information is preserved; on labels, as PDF files accompanying the data files or even as BWAV file headers, an accurate description of any relevant data will prevent mismanagement that leads to bad audio reproduction. Any signal must be decoded in coherence to how it was encoded regardless of whether that encoding is noise reduction, data compression or metadata. Good documentation facilitates proper signal management. Accurately reproducing the audio from a commercial CD is easy. Dealing with thousands of undocumented files in a myriad of formats is extremely difficult. It is good to assume that the work you do will be used by someone who has no idea of your project’s history other than what your notes indicate.

Microphone Technique
The first place to ensure intelligibility is at the very beginning of the signal chain; the microphone. Position is of primary importance. Engineers must strike a balance between two physical characteristics of microphones.

1. The Inverse Square Law. The closer a microphone is to a sound source the louder that source will be in relation to the background sound by an inverse square factor. At a distance of 20 cm a source will be 4 times louder than at a distance of 40 cm. (40÷20=2. 22=4) Sound that emanates from a point source drops in level 6 dB for every 2 units of distance. So a signal of 90dB at 20cm will be 84 dB at 40cm.
2. All directional microphones are subject to the Proximity Effect, where the bass frequencies are increased in relation to the overall tonal balance. See Fig 2.

Therefore if a microphone is too far from a sound source, the source will be indistinguishable from the background noise. If a microphone is too close to a sound source it will suffer from hyperextended low frequency response, making the sound appear muffled. The challenge is to find the position where the microphone distinguishes the desired sound from the background yet does not over-enhance the low frequency components. For most dynamic cardioid mics like the Shure SM 57/58 or Beta 57/58 that distance is generally between 15 and 30 cm.

Frequency Response Curve

The proximity effect can be used to enhance the bass component of a voice. However at distances where the proximity effect becomes apparent a very small adjustment in distance will yield a very large difference in bass response. The proximity effect will also increase the apparent sensitivity to plosives. Certain consonants, especially “P” and “B”, will be over-accentuated.

Wind and vibration are other factors that will affect clarity:
If the microphone is placed in a windy environment or if the talent requires, use a windscreen.
To reduce unwanted sound due to vibration use isolation or shock mounts. Certain microphones are designed for handheld use. Others will always require isolation mounts.
Isolation mounts should be used for orchestras, ensembles, choirs, and sources that are captured from a distance. Large diaphragm microphones also should be used with shock mounts. It also should be noted that in general, the greater the directionality of a microphone the more sensitivity it has to handling and wind noise as well as to proximity effect.

Certain microphones are sensitive to diaphragm overloads. In such cases either move the microphone farther from the sound source or change the microphone. Even extremely durable microphones can develop rigidity in the diaphragm over time, which will make them prone to distortion, as they get older.

Microphone Position
Microphone position is of primary importance regardless of what kind of music or program is being presented. If a microphone is correctly positioned the sound of the instrument will be captured. If a microphone is incorrectly positioned even with a perfect signal chain the sound will be compromised.

Vocals
The human voice was the first musical instrument, so regardless of culture or category of music it is the instrument that bears the greatest scrutiny. As such, artists and producers are most apprehensive and preoccupied with the recording of vocals. Achieving a great vocal sound can be compromised by the performer’s insecurities and/or the producer’s concerns and preconceptions. Clarity in the vocal is objective; you can either hear it or you cannot. The tonal coloration however, is subject to taste, available equipment and how the vocal sound fits within the rest of the tracks (or vice versa) and the remainder of the entire production.

A very important issue for engineers while recording vocals is an overabundance of bass due to the proximity effect mentioned above. Many microphones are equipped with bass filters to ameliorate this problem. Of course when recording an operatic bass this sort of filter cannot be used and the audio operator must rely on positioning. A second common problem is distortion. Distortion is not necessarily obvious nor is it always visible on the meters. Careful listening on an accurate speaker or headphone system is required.

Instruments
Every instrument presents its own particular challenge. Every individual instrument family has general characteristics and individual instruments each have their own sonic personalities.

Some examples:

Woodwinds:
Most of the sound from a woodwind comes through the keys. The sonic character of the instrument, the harmonic overtones and also the fundamental note all radiate from the entire instrument through the keys. Only the fundamental tones and the lowest note emanate from the bell. Flutes are the exception with the majority of the sound coming from the mouthpiece.

Horns develop their sounds from the bell.

Strings form their sounds from the bridge of the instrument. In the case of the violin family, it is a good idea to aim the microphone at both the bridge and the sound post. For guitars, the sound originates from the bridge and radiates outward across the soundboard, or top of the instrument. The sound of the instrument does not emanate from the sound hole. The sound hole acts as a bass relief port similar to the bass port on a speaker.

One of the more interesting challenges is how to make a clean recording of a highly amplified instrument such as a distorted electric guitar. In general, one should be able hear the same sound at the amplifier as through the monitoring system. If the quality of sound remains constant regardless of the monitoring level, all should be well.

Mix balance
Assuming that all the signals are well recorded and free of distortion, clarity can be assured through both level and pan position in a mix.

Equalization is mostly used as a corrective measure. However, EQ can also be used to emphasize or deemphasize components of a mix: In a simple voice over background music mix, a trick used in American radio production is to equalize the music to deemphasize the midrange frequencies so that voice can be heard clearly while the level of the music is still apparent. A small dip (-1.5 - -4 dB) from approximately 500Hz - 3.000Hz in the music track will make an announcer’s voice much more present without sacrificing the level of the music. It is often easier and faster to apply a static process, such as equalization rather than a dynamic process such as ducking, though ducking does work very well and can be quite transparent if set up properly.

Some artists or producers prefer that the main voice or instrument be much louder than any of the other sources. In such cases it is still necessary to pay attention to clarity. A muffled voice will always sound like a muffled voice, even if it is the only thing in the mix.

Dynamics
Compressors, gates and duckers can all be used to enhance clarity of a source. Limiters are considered to be mostly safety devices and are used to prevent the overall level from exceeding a given value, i.e.: -3 dBFS , though they do alter the tonal quality of the signal and as such can be used to change the tonal coloration.

____________________
1 How do Woodwind Insruments Work? J. Wolfe University of New South Wales 1994

Compressors were originally developed to reduce the dynamic range of broadcast and recorded material so it would be coherent with technology available at the time. Transmitters have dynamic range limitations; if fed with a signal that is too loud, the transmitter will be damaged. If the signal is not loud enough then the station’s broadcast coverage area will be attenuated. Originally the audio signal was fed directly into the transmitter. The only level control was a skilled operator who worked with a script or a score to anticipate volume levels and keep the equipment operating at peak efficiency.


The compressor as we know it today has its origins in the CBS Laboratories in New York and Stamford, Connecticut. The Audimax, an automatic gain controller was introduced in 1959 and the Volumax, a peak limiter, was introduced in the 1960’s. Both units were solid state electronic devices. Previous dynamics devices were tube based and were extremely expensive.

Fig 3

Fig 4

Dynamics processors function by using a control or side chain signal to determine the amount of process applied. This control signal is derived from the input or possibly from a parallel signal.
____________________
2 Decibels Relative to Full Scale
3 A short History of Transmission Audio Processing in the United States © 1992 Robert Orban

Fig 5

Compressors have the effect of accentuating whatever frequencies are most prevalent in the signal; Music tends to have more energy in the low frequency regions. Therefore over-compressing a mix will accentuate the low frequencies while masking the high frequencies, making the mix sound dull. However, this characteristic allows a compressor to be used as a de-emphasis device.

Fig 6

By exaggerating the frequencies associated with sibilance at the input to the side chain, the compressor will act on those frequencies and diminish the sibilance. By exaggerating those frequencies associated with plosives at the input to the side chain, the compressor will act on those frequencies and diminish them. Thus can a barrier to intelligibility be overcome. Over-processing can lead to listener fatigue, diminished comprehension and even distortion. An overabundance of any process, whether it is dynamics, equalization or effects can adversely affect the mix. It is absolutely preferable to make corrections early in the process; with the performance, with microphone choice and position, correct level management and so forth. This being so, there are many times when it is not possible to do so. In such cases equalization, dynamics and other processes can be applied as corrective measures. However, being aware of the optimum procedures can lessen the necessity for such corrective measure.

By carefully managing the available resources; talent or event, microphone selection and position, equipment and time, it is possible to maintain intelligibility in the components of any given mix.

Maintaining clarity requires a basic understanding of the physics of sound as applied to microphone technique, familiarity with the available equipment and in general, attention to detail.

Daniel Littwin , director New York Digital.
contact: daniel.nydigital@gmail.com
São Paulo, Brazil

Tags: Column , Daniel Littwin