SONY SDP-EP9ES
¥ 88,000 (released in 1997)
Commentary
This is a Dolby digital processor equipped with digital cinema sound that brings together Sony's digital technologies.
Digital Cinema Sound
It is equipped with Digital Cinema Sound technology, which was proposed taking advantage of the development environment unique to Sony.
The SDP-EP9ES has two modes : Cinema Studio mode and Virtual 3D mode.
Cinema Studio Mode
Cinema Studio Mode is a mode that reproduces the sound of a mixing studio where a sound designer creates the sound of a movie. Three different types of studios are selected from the studios owned by Sony Pictures Entertainment and installed as Cinema Studios A, B and C.
Cinema Studio A reflects the sound characteristics of Cary Grant Theater, Sony Pictures Entertainment's most traditional theater, which has the same volume as a regular medium-sized theater, with a permanent 360 channel input and 72 mix-out console, as well as a state-of-the-art digital hard disk recorder.
In this editing theater, movies by American President and The Fan were edited.
Cinema Studio B is based on the Kim Novak Theater, a Hollywood studio with state-of-the-art audio equipment that supports all currently available output-valued movie sound formats, as well as sound design for CG and other new staging images.
Movies such as Jumanji were edited in this editing theater.
Cinema Studio C reproduces the acoustic characteristics of a scoring studio that records music such as background music of movies.
It has sound field characteristics suitable for viewing musical works and movies with much music, and gone with the wind music was also recorded in this studio.
Virtual 3D Mode
In the virtual 3D mode, the sound field unique to a movie theater can be created more actively by making full use of 3D acoustic technology.
SDP-EP9ES is equipped with four modes : virtual multi-rear mode, virtual rear-shift mode, and virtual enhanced surround mode A/B.
In Virtual Multi-Rear mode, a single set of rear speakers can be used to create a sound field that looks like a large number of rear speakers. This allows you to create a sound field that feels like being wrapped in sound that can only be experienced in a movie theater with multi-rear speakers.
There are two possible locations for the rear speaker. When the rear speaker is placed next to the speaker, the sound field is created as if multiple speakers are placed behind the speaker, and when the rear speaker is placed behind the speaker, the sound field is created as if multiple speakers are placed in front of the speaker.
In virtual rear shift mode, you can enjoy a large scale surround space.
In this mode, the sound source is moved virtually to a non-existent position, so that the sound source is positioned virtually as if it is located past the wall, even if there is a wall. Specifically, you can enjoy the feeling that the room is wider in the back.
The Virtual Enhanced Surround mode allows the rear sound to be positioned to the rear with only the front speaker, and it is very effective when the installation of the rear speaker is difficult.
There are two variations of this mode, A/B. Type A allows you to locate multiple virtual rear sound images in the rear, and type B allows you to locate them in the rear.
Other
In order to reproduce the sound fields of concert halls and live houses, other DSP modes are also installed.
The decoder circuit uses a high-precision 24-bit DSP using the latest process.
This DSP can perform all internal operations with a precision of more than 24 bits (24 bits x24 bits = 56 bits). With an operation capacity of 44 million operations per second, it maintains sufficient precision for decoding Dolby Digital, which repeats complex calculations. This enables sound reproduction taking advantage of the 120 db dynamic range of Dolby Digital.
In addition, this DSP consumes relatively little power and generates relatively little noise, thus minimizing the influence on analog circuits.
Equipped with bus redirection function.
Since l, C, R, SL c, R, SL, and SR signals have a bandwidth of 3 Hz to 20 kHz, so it is ideal to use a speaker with a wide playback bandwidth for all speakers. However, since it is often difficult to install a speaker in reality due to space limitations, a bus redirection function has been formatted so that the low frequencies can be assigned to other speakers so that the system configuration can be adjusted. For example, when a small rear speaker is used, only the low frequencies in the rear can be extracted and assigned to a speaker such as a subwoofer.
The SDP-EP9ES supports all 16 possible combinations of front large / small, center large / small, rear large / small and with / without sub-woofers. In addition, the crossover network processing for low-pass extraction is done in the DSP with 24-bit digital processing at the digital stage to minimize sound quality degradation.
The newly developed digital i-volume is mounted on the volume.
The digital i-volume is designed to control the current value of the reference current source of the current pulse D/A converter mounted as a D/A converter with the output of a static D/A controlled by a microcomputer. This allows the overall volume to be adjusted arbitrarily while maintaining the resolution of the digital data. Not only the overall volume but also the balance adjustment is processed in the digital domain.
The volume on the front panel is connected to a microcomputer for volume position detection, and there is no analog volume on the audio line, including the volume for balance adjustment between channels.
Equipped with a jitter-free D/A converter system.
A digital signal processor such as a processor extracts a clock signal sent from a source device, generates a master clock in synchronization with the extracted clock signal, and operates a D/A converter. However, the clock signal sent from the source device may contain jitter, which adversely affects sound quality. In the past, an analog PLL circuit was used to extract the clock signal and compensate for jitter in order to reduce this effect. However, even if the compensation is performed, the signal basically follows the input signal.
In order to solve this problem, the SDP-EP9ES separates the input clock and the master clock of the D/A converter. As a result, the D/A converter operates on its own master clock and is free from jitter. By performing stable D/A conversion, excellent sound quality is obtained.
All six channels, including the subwoofer, use circuits of the same quality.
The D/A converter employs a full feed-forward digital filter and 6-channel independent current-pulse D/A converter. In order to achieve high-quality D/A conversion in the SDP-EP9ES, the D/A converter occupies approximately two-thirds of the main circuit board.
In digital filters, the first complementary data is generated at an intermediate point between the data obtained from the disk and the data, and the second complementary data is generated using the generated data. The digital filters are connected in three stages. However, the generated complementary data contains much finer values than the original data than the original data, so if it is passed to the next stage as it is, the circuit scale becomes large. In conventional digital filters, the low-order bits of the calculation result are rounded off and only the high-order bits are passed to the next stage. Therefore, requantization noise occurs due to rounding errors.
In the full feed-forward method digital filter, the rounded lower bits are transmitted by another route, stored in memory, and combined with the result of the next stage operation. By performing full feed-forward operation, the over-sampling operation that also reflects the lower bit data is realized, thereby almost completely eliminating the requantization noise within the audible band.
The Pulse D/A converter converts the input digital data into a high-density pulse train and passes the resulting pulse train through a low-pass filter to reproduce the audio signal. However, since the voltage value representing the height of the pulse train is the power supply voltage applied to the D/A converter IC, fluctuations in the power supply voltage affect the pulse output. Moreover, noise generated by the digital arithmetic circuit inside the D/A converter appears as noise in the pulse output and affects sound quality.
In the current pulse D/A converter, a stable and clean current source is prepared first, and a current pulse is generated by switching using the output of the pulse generator. The current pulse thus obtained is free from voltage fluctuations and is not affected by the voltage noise of the arithmetic circuit.
Each of the front, center and rear channels has its own digital tone controller that allows you to adjust the high and low tone in the digital domain. In addition, you can set the turnover frequency to any point in the extended menu mode. There is also a base boost function that increases the impact of the low tone with one touch.
These processes are performed by a 24-bit precision DSP similar to the decoding of Dolby Digital.
A bypass input terminal is mounted.
When the power supply is OFF or when BYPASS is selected with the input selector, the signal of the device connected to the bypass input terminal can be output as it is.
A FB chassis is used for the chassis.
The FB chassis is made of metal material with sufficient strength and thickness, and the entire chassis is firmly fixed as one box by the frame (Frame) and the beam (Beam) that is passed to the left and right (Beam). In addition, members with different natural vibration modes are combined by using different shapes and sizes to cancel each other natural vibration and greatly reduce the vibration of the entire chassis.
The internal beam also acts as a shield that separates the power transformer from the circuit board, preventing interference with the audio signal.
An eccentric insulator is used for the legs.
This insulator has a screw hole that avoids the center position, changing the phase of the transmitted vibration, and canceling it makes it difficult for the vibration to be transmitted to the chassis.
The power transformer is quite large for a processor which is a low-power device.
Most operations can be performed with two rotary encoders, "Menu" and "+ / -", to improve operability.
Wireless remote control is included.
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Model Rating
| Type | Dolby Digital Processor |
| Surround Mode | Refer to the surround mode list below |
| Dolby Pro Logic Input Balance | Auto |
| Tone control | Front, center and rear independent Bass : ± 10 dB (center frequency 99 to 992 Hz) Treble : ± 10 dB (center frequency 1.0k to 8.6 kHz) Increment / decrement step : 0.5 dB |
| Bus boost | +5dB(60Hz) |
| Digital input | Light : Route 3 Coaxial : 1 system AC-3RF : 1 System |
| Digital output | Light : 1 system |
| By-pass input terminal | Route 1 |
| Audio Out | Front L/R : 1 system Center out : 2 routes Rear L/R : 1 system Subwoofer : Two lines |
| Pwer | 100 VAC, 50Hz/60Hz |
| Power consumption | 40W |
| External dimensions | Width 430x Height 98x Depth 355.5 mm |
| Weight | Approx. 6.5 kg |
| Attachment | Wireless Remote Control RM-EP9 |
| Surround Mode | Effect obtained |
| Dolby Pro Logic | Dolby It is used to play back a program processed by pro-logic. It is used when pro-logic processing is performed on the 2 ch source. |
| Dolby Enhanced Pro Logic | When the rear is monaural, such as Pro-Logic, it increases the sense of presence. This function performs pseudo-stereo processing on the rear channel. |
| Large Theater | Simulates the sound characteristics of a typical movie theater. |
| Cinema Studio A | It reproduces the sound characteristics of the mixing studio "Cary Grant Theater" and is a standard movie viewing mode. |
| Cinema Studio B | Reproduces the sound characteristics of the Kim Novak Theater mixing studio, suitable for watching SF-X and action movies with a lot of sound effects. |
| Cinema Studio C | Reproduces the acoustic characteristics of a scoring stage for music recording. Suitable for listening to classical music and musical film. |
| Virtual Multi Rear (A / B) | Creates sound as if there are multiple rear speakers. Unfolds the virtual sound source to the rear (A) or side (B). |
| Virtual Rear Shift (A & B) | Shift the position of the rear speaker backward (A) or laterally outward (B). |
| Virtual Enhanced Surround A | Only the front speaker can locate multiple virtual rear sound images in the rear. |
| Virtual Enhanced Surround B | Two virtual rear sound images are positioned behind the front speaker only. |
| Large Hall | Reproduces the sound field of a concert hall for classical music, etc. |
| Live House | It reproduces the sound field of a live house for a live concert. |
| * can be adjusted when the menu mode is in the extended mode. * can be adjusted only in the basic mode. | ||
| Parameter Name | Menu encoder | + / - encoder |
| SP.SETUP | Front SP(front speaker size) | Large/Small |
| Center SP(Center Speaker Size) | Large/Small/No | |
| Rear SP(rear speaker size) | Large/Small/No, Behind/Side | |
| Sub Woofer(with or without a subwoofer) | Yes/No | |
| Front(Distance between the front speaker and the viewing position) | 1 ~ 12 meter <0.1m step> | |
| Center(Distance between Center Speaker and Viewing Position) | 0 ~ 12 meter <front -1.5m ~ front> | |
| Rear(Distance between Rear Speaker and Viewing Position) | 0 ~ 12 meter <front -4.5m ~ front> | |
| *Front SP(front speaker cutoff frequency) | Small = 60 Hz ~ 200 Hz <60 / 80 / 100 / 120 / 150 / 200 Hz> | |
| *Center SPCenter Speaker Cutoff Frequency | Small = 60 Hz ~ 200 Hz <60 / 80 / 100 / 120 / 150 / 200 Hz> | |
| *Rear SP(rear speaker cut-off frequency) | Small = 60 Hz ~ 200 Hz <60 / 80 / 100 / 120 / 150 / 200 Hz> | |
| Level Adjust | Test Tone(Balance Adjustment Test Tone) | Auto → L → C → R → SR → SL/Off |
| Front(front speaker balance) | ± 8.0 dB <0.5 dB step> | |
| Rear(rear speaker balance) | ± 8.0 dB <0.5 dB step> | |
| Rear Level(rear speaker level) | -20.0 db to + 10.0 db <0.5 db steps> / Mute | |
| Center Level(center speaker level) | -20.0 db to + 10.0 db <0.5 db steps> / Mute | |
| Sub Woofer(subwoofer level) | -20.0 db to + 10.0 db <0.5 db steps> / Mute | |
| Surround | Surr.Effect(Surround Effect Level) | 0% to 100% <5% steps> |
| LFE Mix.(Heavy Bass Mix Level) | -20.0 db to + 0 db <0.5 db steps> / Mute | |
| D.Range CompDynamic Range Compression Level | OFF/0.1 ~ 0.9 <xxx Steps> / STD/MAX | |
| *Tone Control | ON/OFF | |
| *Bass | ± 10 dB | |
| *Trble | ± 10 dB | |
| Equalizer | *Equalizer | ON/OFF |
| *Front Bass | 99 Hz ~ 992 Hz <21 points>, ± 10 dB <0.5 dB steps> | |
| *Front Treble | 1.0 kHz to 8.6 kHz <21 points>, ± 10 dB <0.5 dB steps> | |
| *Center Bass | 99 Hz ~ 992 Hz <21 points>, ± 10 dB <0.5 dB steps> | |
| *Center Treble | 1.0 kHz to 8.6 kHz <21 points>, ± 10 dB <0.5 dB steps> | |
| *Rear Bass | 99 Hz ~ 992 Hz <21 points>, ± 10 dB <0.5 dB steps> | |
| *Rear Treble | 1.0 kHz to 8.6 kHz <21 points>, ± 10 dB <0.5 dB steps> | |
| Costomize | Muting | ON/OFF |
| Disp.Dimmer | 25% to 100% <1% steps> | |
| Dist.Unit(Distance Units) | Meter/Feet | |
| EQ Memory | ON/OFF | |
| Memory Clear | Yes/No | |
| Menu Mode | Basic/Expand | |
| *Input Trim(Digital Input Ratio Adjustment) | + 5.0 dB ~ -5.0 dB, ± 10 dB <0.5 dB step> | |
| *Decode Mode(Digital Input Decode Mode) | Auto/AC-3 | |
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