DESCRIPTION

The present invention relates to a method and system for altering a supplementary audio recording, for adding to a video recording.

BACKGROUND OF THE INVENTION

When a film or video of a scene is shot using a camera and a microphone, the audio from the microphone is recorded in syncronisation with the picture from the camera. This is known as 'Production Audio'. However, circumstances can and often do arise where the production audio recordings are unusable. These may be anything from an aircraft flying overhead during the shooting of an historical scene or quite simply an inadequate performance.

At present, the way to replace this audio with more suitable audio is by using three basic processes, namely ADR, Foleys and Sound Effects Editing. ADR (Automated Dialogue Replacement) is where an actor will go into a recording studio after the filming is over and repeat the line or lines he has performed in the scene, usually one by one. The scene is projected on a screen while the new performance is recorded so that the director and sound editor can gauge whether or not the performance is sufficiently in sync with the picture.

Foley is where a Foley artist will re-record the actor's action in sync with the picture; the footsteps, body movements, the sound of the actor sitting on a chair, picking up and putting down a cup or phone - practically everything the actor does. All these recordings are then given to the sound editors to put into the soundtrack and to 'fit' or cut into sync. Sound editors will usually then add sound effects from their archives or from recordings made during the shoot (known as 'Wild Tracks')

This is then taken into a pre-mixing theatre where the re-recording engineer will try to make these edited recordings sound as close to the original Production Audio recordings as possible.

There are many problems which arise using this method, but they mostly relate to the simple fact that Post Production Sound Departments find it very difficult to match the Production Audio recordings with complete accuracy. The equipment setup and recording environment used in production is may be very difficult to replicate well enough for the re-recordings to sound like the original production recording should have sounded. Also different ADR studios (sometimes in different countries) may be used to record actors playing in the same scene meaning the microphones may not only be different from the original Production microphone but also different from each other.

Another major problem that can arise from the Production audio recording is the proximity effect caused by the use of Radio Microphones which are typically positioned directly on the actor to be recorded. Therefore, the distance of the actor from the camera is not reflected in the audio recording, as the microphone is too close to them. The re-recording engineer must then manually recreate this space or perspective to make the audio sound natural and realistic.

It is therefore an object of the invention to provide an improved method and system for altering supplementary audio recordings for adding to video recordings.

SUMMARY OF THE INVENTION

The present invention recognises that one of the reasons that production audio is so hard to replicate afterwards in the studio is that the distance between the actor and the microphone/camera may be changing from frame-to-frame as the actor speaks, such that there is no single one set of changes that can be applied to re-recorded audio to make it match the production audio. As the actor moves from one part of the scene to another part of the scene whilst speaking a line, the way the line sounds will change.

One component of the way in which the line changes when the line is incorporated into a stereo audio track in post production, is the panoramic positioning of the audio in the soundscape. The sound from the relevant microphone may be mixed into the stereo audio tracks to make it sound like the actor is moving across the scene.

However, this is a separate issue to being able to correctly reproduce the production audio that is recorded by a given microphone during filming. As the actor moves through the scene, the distance and direction from the actor to the microphone/camera may change, making the actor sound progressively louder/quieter, and/or differ in frequency according to the frequency response of the microphone, and the amount of reverberation may change depending on whether the actor is speaking towards an open space or an enclosed space. Such changes can be independent of whereabouts the actor's voice is to be positioned in a stereo soundscape, and cannot be compensated for by simply choosing whereabouts to place the sound recording in a panoramic soundscape. According to a first aspect of the invention, there is provided a method of altering a supplementary audio recording, in preparation for adding to a video recording of a scene comprising a sound source. The video recording was recorded by a camera and a microphone, and the supplementary audio recording was recorded at a different time to the video recording. The method comprises receiving the supplementary audio recording and location information defining relative positions of the sound source and the microphone, or relative positions of the sound source and the camera, and altering characteristics of the supplementary audio recording based on the relative positions.

Accordingly, the supplementary audio recording may be altered to replicate the sound that would have been received at the microphone in the production environment, had the production audio recording not been compromised. The sound source does not have to be an actor, but could for example be an object clashing with another object, etc. The distance from the sound source to the microphone may be augmented with the distance from the sound source to the camera, so that the sound is altered to give a viewer of the video a perception that the sound is with reference to their position, i.e. at the camera.

The supplementary audio is typically intended to replicate the sound that would have been recorded by the microphone in the production environment, and so the supplementary audio recording is typically a monaural audio recording, i.e. it only has one audio channel, as opposed to stereo which normally has left and right audio channels.

The method may automatically and completely match the supplementary audio recorded in the Post Production process to the Production Audio that it is intended to replace, by frame- accurately simulating the original technical and acoustical conditions of the Production Audio recordings.

Foreign dubs of a film use a music and effects track (M&E) where the original production sound of the film is used as much as possible. The present invention can automatically enable the dubbed voices and additional Foleys to sound exactly as they would have done had they been recorded on the original location, thus matching the original production sound. The supplementary audio may be added to the video recording to produce an altered video recording. The altered video recording may be used by recording it to a recording medium, broadcasting or otherwise distributing, and/or incorporating it into another video recording. The method may further comprise receiving a characterization of the microphone, the characterization specifying at least one of the sensitivity and frequency response of the microphone over a range of distances and angles from the microphone, wherein the altering of the characteristics of the supplementary audio recording is based on the characterization. Then, the supplementary audio recording may be made to sound like it was recorded by the production microphone, so that it sounds correct compared to other sounds recorded by the production microphone that are retained in the video recording rather than being replaced.

The relative positions of the microphone and the sound source may indicate whether the sound source is behind the microphone, i.e. if the microphone is pointing in a direction greater than 90 degrees away from where the sound source is located. If the sound source is behind the microphone, then an amplitude and/or equalisation of the supplementary audio recording may be modified differently than to when the sound source is in front of the microphone.

The location information may further comprise an audio impulse response of the scene, and altering the characteristics of the supplementary audio recording comprises convoluting the supplementary audio recording with the audio impulse response of the scene. Then, the supplementary audio recording can be made to sound as though it was actually recorded in the production environment, rather than in a studio. Advantageously, the altering of the characteristics of the supplementary audio recording may comprise increasing a reverberation of the supplementary audio recording as the relative positions of the sound source and the camera indicate an increasing distance between the sound source and the camera. Then, the supplementary audio recording may sound more realistic as it correctly reflects what is happening to the sound as heard by the viewer at the camera.

Since the supplementary audio is only a single audio channel from the single microphone where it would have been recorded in the production environment, after the supplementary audio has been altered to sound as though it was recorded in the production environment, the altered supplementary audio may be processed to generate a plurality of audio tracks for respective speakers in a multi-speaker audio reproduction system. The plurality of audio tracks panoramically position the supplementary audio recording within a soundscape. The processing of the audio tracks may comprise changing the panoramic position of the altered supplementary audio recording as the relative positions of the sound source and the camera indicate a changing angular position of the sound source to the camera. The method may further comprising filming the scene with the camera and the microphone, and recording the location information during the filming. The term "filming" is not restricted to filming with film, and includes filming with digital cameras.

According to a second aspect of the invention, there is provided a system for processing a supplementary audio recording, in preparation for adding to a video recording of a scene comprising a sound source. The video recording was recorded by a camera and a microphone, and the supplementary audio recording was recorded at a different time to the video recording, and the system comprises:

an audio input for receiving the supplementary audio recording;

a location information input for receiving location information defining relative positions of the sound source and the microphone, or relative positions of the sound source and the camera; and

a microphone modeller configured to alter characteristics of the supplementary audio recording based on the relative positions.

The system may further comprise a deconvolutor that is configured to alter the characteristics of the supplementary audio recording by deconvoluting the supplementary audio recording with an audio impulse response of a recording environment in which the supplementary audio recording was recorded. Accordingly, if the supplementary audio was recorded in a small room, then the effects of the small room on the audio can be removed from the audio. The effects of the production environment may then be added to the audio by convolving the audio with an audio impulse response of the production environment (the scene).

A third aspect of the invention provides an apparatus for recording a video recording of a scene and location information, to provide the video recording and location information. The system comprises a camera, a microphone fitted with a position detecting device, an ancillary position detecting device for fitting to a sound source of the scene, and a transceiver for receiving detected positions from the position detecting devices. The position detecting devices may for example be GPS devices, and allow the relative positions of the microphone and/or camera to be determined with respect to the sound source during the filming of the screen. Fourth and fifth aspects of the invention provide a method and apparatus for calibrating a microphone, so that supplementary audio recordings can be recorded using the microphone when a calibrated microphone is unavailable. A microphone calibrator is itself calibrated using a calibrated microphone and a signal generator, and then the microphone calibrator and signal generator can be used to calibrate uncalibrated microphones.

The term "video recording" is sometimes used in the art to mean only moving pictures, with any accompanying audio considered to be a separate component. However, as used herein, the term "video recording" encompasses both moving pictures and any accompanying audio which may or may not be present, and as such the video recording could alternatively be described as an audio- visual (AV) recording.

DETAILED DESCRIPTION Embodiments of the invention will now be described by way of non-limiting example only and with reference to the accompanying drawings, in which:

Fig. 1 shows a schematic diagram of a system for supplementing a video recording with supplementary audio, according to an embodiment of the invention;

Fig. 2 shows a schematic diagram of a microphone recalibrator, for use in embodiments of the invention;

Fig. 3 shows a schematic diagram of a camera having reference transceivers in accordance with an alternate embodiment of the invention;

Fig. 4 shows a schematic diagram of a camera having reference transceivers in accordance with another alternate embodiment of the invention; and

Fig. 5 shows a schematic diagram of panoramically positioning the supplementary audio in a soundscape.

The figures are not to scale, and same or similar reference signs denote same or similar figures.

A first embodiment of the invention will now be described with reference to Fig. 1. Fig. 1 shows a recording apparatus 100 incorporating various position detecting devices Tl, T2, T3, T4, T5, TL, TR, TN in the form of GPS transceivers that are used to calculate the relative distances and angles of camera C, microphone E, actor F and object(s) within a frame of reference defined by the reference GPS transceivers Tl and T2. The method used to measure the distance of each transceiver from the others may alternatively be Ultrasound, Infrared, Laser, Magnetic, Bluetooth, or any other position measuring technique. For example, one possibility would be to use AHRS inertial navigation modules, as will be apparent to those skilled in the art. Then, height information could additionally be recorded. The angles between the various transceivers may be calculated by either using triangulation, or by using the transceivers' orientation relative to a fixed reference direction, such as Magnetic North. Or, a Virtual North may be defined as being the location of the positioning detecting device of the camera, and the angle of each of the other position detecting devices may be defined as the angle between the direction in which the position detecting device is pointing and the direction towards the Virtual North, specifically the direction towards the position detecting device at the camera.

In this configuration transceivers Tl and T2 are the dedicated base (reference) transceivers from which the encoder B calculates the relative distance and angle of the other transceivers. These transceivers may be static when shooting conditions permit. Should conditions not permit this, such as when the camera is handheld and free moving along with the sound recordist, transceivers Tl and T2 are attached to the cameraman and to the sound recording equipment respectively. For example, where the cameraman is shooting alone with the audio being recorded by a microphone mounted on the camera, rather than separately, the reference transceivers Tl and T2 may be mounted at a distance from each other onto the camera by use of a bar mounting (Figure 3), or an aerial-shaped mounting (Figure 4). Alternatively, the reference transceivers Tl and T2 may be excluded, and each transceiver may be equipped with a digital compass, which relays the transceivers' orientation relative to magnetic north to the encoder (B), along with their relative distances to each other.

The transceiver T3 of the camera may be used to define the Virtual North as the location of the camera. Then, the position detecting device of each transceiver can be used to determine the direction from the position detecting device towards Virtual North, and the compass of the transceiver can be used to detect which direction the transceiver is pointing in relative to the direction towards Virtual North.

The encoder B receives wireless signals from the position detecting devices Tl, T2, T3, T4, T5, TL, TR, TN, and outputs positional data indicating their relative positions. The encoder B is connected to the camera C, and receives a sync reference/timecode by means of a BNC cable connected to the timecode output of the camera, or, whenever possible, wirelessly. The sync reference/timecode is also sent to the digital audio recorder D that is connected to the microphone E for recording the production sound. The sync reference/timecode may also be connected to the camera's sync break out box when circumstances permit. These data are then calculated and combined to form a code or data stream logging the positional information of each transceiver for each frame of the film or video.

This data stream is then encoded (hereinafter called the 'CASPAR Code') and recorded either separately or as metadata imbedded within the production audio recording. A safety backup is simultaneously made to a removable SSD (or other removable medium) mounted in the SSD (or other removable medium) drive G housed within the encoder B.

An IR (Impulse Response) Sweep of the location/camera set-up is made and the resultant time domain artifact data recorded is on the digital audio recorder D prior to shooting the scene. All lighting and other equipment must be in place before this is made. The digital audio recorder D outputs the IR Sweep for later processing.

The filming of the scene having the actor F is then performed, and audio from the actor F is recorded by the fixed position microphone E as the actor moves around the scene. The transceiver T5 tracks the movement of the actor around the scene, and the encoder B receives and stores the relative positions of the actor F, microphone E, and camera C throughout the filming of the scene, and outputs this as positional data. Unfortunately, some of the audio from the actor F is corrupted by sound from a plane flying overhead, and will need to be replaced in such a way that it matches in with the rest of the audio from the actor F that was recorded during the scene.

Therefore, the Actor F or a Foley Artist E2 re-performs lines or recreates action in a controlled environment 200, and these are recorded and sent to the CASPAR Controller 300.

In this embodiment, the lines or action performed by the Actor F or a Foley Artist E2 in the controlled environment are recorded in sync with the picture from the camera C using a calibrated microphone D2, to ensure the signal is as transparent or 'flat' as possible, as will be apparent to those skilled in the art. However, in an alternate embodiment where a calibrated microphone is unavailable, the microphone recalibrator 500 may be used, as will now be explained with reference to Fig. 2. Fig. 2 shows a signal generator 400, which is a unit encasing a signal/pink noise generator B4, an amplifier C4 and a loudspeaker D4. It can be mounted on a standard microphone stand using a threaded base-mounted housing able to accommodate both 9.5 mm (3/8") and 15.9 mm (5/8") fittings.

The microphone recalibrator 500 is a software plugin or hardware unit comprising, connected in sequence; a microphone input, an automatic variable attenuator G5, a spectrum analyzer H5 and an EQ programmer J5.

The microphone recalibrator 500 is itself first calibrated by sending Pink Noise and a sine sweep in an anechoic environment through a calibrated scientific microphone using the signal generator 400. The SPL (amplitude) and frequency response data of the calibrated scientific microphone are then saved as DEFAULT DATA against which uncalibrated microphones are subsequently measured.

To calibrate an uncalibrated studio microphone E4, the signal generator 400 is placed so that the loudspeaker D4 is directly facing the studio microphone (E) at a distance of 1 meter. Then, the rotary dial selector A4 is set to pink noise, and the pink noise is played into the studio microphone E4 through the loudspeaker D4. The resulting signal from the studio microphone E4 is sent through the standard studio signal path F4 into the microphone recalibrator 500.

The automatic variable attenuator G5 of the microphone recalibrator 500 receives the signal from the standard studio signal path F4, and adjusts the input level so that it matches that of the DEFAULT DATA.

Next, the rotary dial selector A4 is set to sine sweep. The sine sweep is played into the studio microphone E4 through the loudspeaker D4 and sent through the standard studio signal path F4 into the microphone recalibrator 500. A spectrum analyzer H5 of the microphone recalibrator 500 is connected to the output of the automatic variable attenuator G5, and receives the attenuated signal from the studio microphone.

The spectrum analyser H5 analyses the frequency response of the studio microphone. This frequency response is then used by the EQ Programmer J5 to create an EQ that makes the frequency response match that of the DEFAULT DATA that was recorded when using the calibrated microphone. The attenuation and EQ settings can then be saved for future use in a memory K5.

The microphone recalibrator 500 can be used to 'flatten' or make transparent a recorded signal from a radio microphone or other undesired microphone. Furthermore, the microphone recalibrator 500 could be used to remove a proximity effect from the production audio, caused either by the signal source being too close to the microphone or by the use of Radio Microphones.

Referring again to Fig. 1, the audio performed by the Actor F or a Foley Artist E2 in the controlled environment 200 and captured by the microphone B2 or D2 is then sent to the CASPAR Controller 300.

The audio recorded in the controlled environment 200 is placed in sync with the production audio that carries the embedded CASPAR Code, or is otherwise synchronized with the CASPAR Code, so that the CASPAR Code controls the digital audio devices contained within the CASPAR Controller 300 in sync with the film or video. The CASPAR Controller 300 processes the audio such that it matches the original production audio it is intended to replace.

The CASPAR Controller 300 is a software plugin or hardware unit comprising in sequence; a deconvolution unit A3, a microphone modeler B3 and a convolution reverb unit C3. The CASPAR Code controls the parameters of the microphone modeller and convolution reverb unit in real time, thus simulating the technical and acoustical conditions of the Production audio recordings. The processed audio is sent to the Premix Theatre. The deconvolution unit A3 is only used in cases where a replacement audio performance by the actor has been recorded in an uncontrolled acoustic environment away from the original film location such as a hotel room or Winnebago. In this case an Impulse Response sweep of the uncontrolled environment is made as reference data for the deconvolution unit, and the deconvolution unit deconvolutes the replacement audio performance to remove the effects of the uncontrolled environment on the recording. Otherwise, if the replacement performance was recorded in a controlled environment such as the studio 200, then the deconvolution unit may be bypassed.

The microphone modeller B3 is intended to add the effects of the production microphone to the replacement audio that was recorded in the studio 200. Then, the replacement audio will sound like it was recorded by the production microphone, rather than the studio microphone, to properly match the rest of the audio of the recorded video.

The microphone modeller B3 is controlled by a mic modeller parameters unit D3, which allows selection of the microphone that was used to record the production audio. There is a minimum of eight selectors enabling simulation of a recording environment where more than one microphone has been used, be they the same or different models to one another.

The parameters being controlled in the microphone modeler are the transceiver router, microphone model selector, amplitude and variable EQ. The transceiver router enables the operator to manually select a specific production microphone by means of its transceiver ID and route the audio recorded with it to one of the microphone models. The microphone model selector is manually controlled.

The amplitude is controlled by the relative distance from T4 to T5, between the microphone E and actor F in the production environment (see Fig. 1), as specified in the CASPER code that is received by the CASPER controller 300. When the distance T4T5 < 1 meter,

Amplitude = 1 (unity gain). Where T4T5 > 1 meter, this value decreases proportionally.

Accordingly, the volume is reduced as the actor F moves further away from the microphone E.

The increment by which the increase or decrease in amplitude relates to T4T5 is defined by the technical specs released by the manufacturer of the microphone model selected, or by independent scientific measurement of the microphone model.

Where the CASPER code indicates that the actor F was behind the microphone, i.e. located greater than 90 degrees from a direction in which the microphone points, then the increase or decrease in amplitude related to T4T5 is defined by independent scientific measurement at increments of a minimum 8cm ~ 3".

The variable EQ is also controlled by the distance T4T5, and the EQ response of each modelled microphone is defined by the technical specs released by the manufacturer of the model selected, or by independent scientific measurement. In the case of the actor being behind the microphone, the EQ response of each modelled microphone is defined by independent scientific measurement at increments of a minimum 8 cm ~ 3".

The convolution reverb C3 is intended to add the effects of the production environment to the replacement audio, so that the audio sounds like it was recorded in the production environment, rather than in the studio or an uncontrolled environment. The convolution reverb C3 is controlled by a convolution reverb parameters unit E3.

The parameters being controlled are reverb program, reverb wet/dry balance and early reflections. The reverb program is dictated by the IR sweep that was taken in the production environment (as described earlier), and that is received by the CASPER controller 300 and loaded into the convolution reverb unit C3. Convoluting the replacement audio with the IR sweep of the production environment adds the effect of the production environment to the replacement audio.

The reverb wet/dry balance is controlled by the relative distance from T3 to T5, between the camera C and actor F in the production environment, as specified in the CASPER code that is received by the CASPER controller 300. The wet/dry balance controls how strongly the effects of the production environment are added to the replacement audio, wet being the strongest (most reverb), and dry being the weakest (least reverb), as will be apparent to those skilled in the art. When the distance T3T5 < 1 meter, wet = 0 and dry = 100. Where the distance T3T5 > 1 meter, the value for wet proportionally increases and the value for dry proportionally decreases the greater the distance T3T5. These values are directly proportional to the amplitude curve of each modelled microphone as defined in the mic modeler parameters unit D3. This parameter control is disabled by default and can be selected by the operator. Early reflections are controlled differently, depending on the number of channels contained in the recording of the IR Sweep. In a stereo reverb (where the IR sweep has been recorded with 90° crosspaired microphones or similar phase-conscious microphone configurations), early reflections are controlled by the left/ right positioning of T5. The delay time of early reflections summed to the left decreases as T5 moves left, while the delay time of the early reflections summed to the right increases; and vice versa.

In a 5 -Channel Reverb (where the IR sweep has been recorded with a 5 -mike configuration; left, centre, right, left surround and right Surround), the distance T3T5 is combined with the panoramic positioning of T5. In this configuration, the CASPAR Code controls the delay times of the early reflections appearing left+right and front+rear using the same principle described in the Stereo configuration above but incorporating the angles shown in Fig. 5 where 90° = front early reflections and 270° = rear early reflections.

The CASPER controller 300 further comprises a global parameters unit F3 at the output of the convolution reverb unit C3, which can optionally be used to panoramically position the replacement audio recording within a soundscape of a plurality of audio tracks. Referring to Fig. 5, it is assumed the camera points at an angle of 90° from Tl, i.e. the default value of T3 = 90°. In a 2-channel stereo environment, the dedicated base transceivers Tl and T2 are considered stereo left and stereo right respectively, and have directional values where Tl = 0° and T2 = 180°. In a 5-Channel environment, values from 0° - 180° are Front Speaker Values whereby: 0° = LEFT, 90° = CENTRE, 180° = RIGHT. The transceivers TL and TR) may also be placed on the left edge and right edge of the camera's Lens Hood. The left edge will denote edge of screen left (65% left) and the right edge will denote edge of screen right (65% right). By use of subtractive calculation, the L-R Front Stereo image is 'pulled in' so that: edge of screen left (TL) = 58.5°, and edge of screen right (TR) = 148.5°

TL and TR are determined by the lens width, which is factored in by using the lens selector Bl on the encoder B. The on-screen L-R stereo position of the object or actor is calculable using T3T5 relative to the width of view of the lens. In the case of 8-Channel systems such as SDDS, LEFT CENTRE = 45°, RIGHT CENTRE = 135°.

Values above 180° and below 360° are Rear Speaker Values whereby: 181° = RIGHT SURROUND, 270° = SURROUND CENTRE, 359° = LEFT SURROUND. Automatic positioning within the monitoring of the Multi-Channel environment will be calculated thus:

Front LEFT, CENTRE and RIGHT: All values between 0° and 180° will be considered

Front Speaker values such that the balance between Front and Rear is FRONT = 100, REAR = 0.

LEFT, CENTRE RIGHT, LEFT SURROUND, RIGHT SURROUND: Values greater than 180° and less than 360° are considered rear speaker values. In this case the AMPLITUDE DATA of the signal relative to the DEFAULT DATA is used to control the balance between front and rear speakers. As the Amplitude decreases the balance moves from Front to Rear proportionally, i.e. the Amplitude of the signal in the REAR speakers (Behind camera) is inversely proportional to the AMPLITUDE DATA relative to the DEFAULT DATA whereas the Amplitude of the signal in the FRONT (In front of Camera) is proportional to the AMPLITUDE DATA relative to the DEFAULT DATA.

The CASPER code may additionally include height information sensed by each of the position detecting devices. Then, sound adjustments can also be based on the relative heights of the sound source, microphone, and camera. Furthermore, in a speaker environment having speakers located at different heights to one another, the height information in the CASPER code may be used to panoramically position sounds in a vertical plane, as well as in a horizontal plane. In further applications of the invention, a proximity effect, caused either by the signal source being too close to the microphone or by the use of radio microphones can be eliminated by sending the original production recording system through the microphone recalibrator in to the microphone modeler. A default distance of T1T5 = 1 Meter is applied, simulating the optimum position of a boom microphone, thus creating a 'Ghost Boom'.

An effect of perceived distance can be added to the supplementary audio by using the CASPAR Data in the microphone modeller B3 to replace T4T5, or gradually add to T4T5, the distance T3T5, thus 'placing' the microphone in the same position as the camera. Where a wide angle or fish-eye lens is being used, the perceived distance from the camera to the subject is greater. By providing the encoder B with camera lens data from the lens selector Bl, a perceived distance may be simultaneously incorporated into the calculations of the encoder.

Off-mic signals are created when the actor turns away from the microphone while speaking. This can be matched by CASPAR by smoothly morphing from the Amplitude and EQ values of the modeled microphone at the time the actor starts speaking to its opposite values behind the microphone at the time the actor has fully turned away. For example, referring to Fig. 5, if T4T5 = 1 meter at 45° the off-mic signal = 1 meter at 315°. The time this takes is programmable by the operator using the timecode reference in the picture.

Automatic archive / wildtrack FX panning can be achieved by placing the desired effect in sync with the production audio that carries the embedded CASPAR Code, or otherwise synching to the CASPAR Code, so that the CASPAR Code controls the digital audio devices contained within the CASPAR Controller 300 in sync with the film or video. An example is a car which has had transceivers placed at its front and rear during the shooting of the scene. A static snapshot is taken of the positions of all transceivers at the beginning of the scene. The panoramic information can then be used for any wildtrack ambience that may have been shot for the scene. It may also be applied to archive ambiences. Many other variations of the described embodiments falling within the scope of the invention will be apparent to those skilled in the art.