The ocean is where life began and evolved.
It is said that all life forms were born in the ocean and some of them
returned to the ocean at the end of the cycle of their evolution.
The concept of the NT series is to be the starting point of professional creative work

Tamura Resource Network Technologythat can constantly evolve and to continue to be like the ocean, a place for evolution.

Simplified connections between units
A single-mode optical fiber cable is used for TR-LINK.
In addition to 512-channel audio signals, synchronousand control signals are transmitted via a single optical fibercable.
Therefore, the synchronous signal cable and control signal
cable, which were conventionally required for each unit
along with the audio cable, are no longer required, and theunits are connected to each other using a pair of opticalfiber cables only.

32-bit floating point data transmission
MADI was previously used for connections between theI/O unit and the audio processing unit.
MADI performs 24-bit fixed point data transmissions,however, and even if the DSP core performs
high-precision arithmetic operation, some data loss isunavoidable owing to data transmission using MADI.
When using TR-LINK, on the other hand, all audio data
are transmitted in a 32-bit floating point data state.
Thus, as long as the I/O unit is connected to DSP via
TR-LINK, regardless of the distance between them, there
will be no data loss at all, as if these units are connected
in a single cabinet.
Analog audio signals to be input to the I/O unit areconverted into 32-bit signals in the I/O unit, whereas
analog audio signals to be output from the Line-out card
are directly converted into analog audio signals from32-bit signals in the I/O unit.

Separation of units

Easy maintenance
Maintenance of the router unit and DSP core, which form
the heart of the system, is performed by unit-wise
replacement instead of the more troublesome
replacement of a circuit board.
Because connections between the units are made using
optical fiber cables only, a faulty unit can be replaced evenwhen the system is operating. Connecting anddisconnecting an optical fiber cable while the system isoperating does not affect the system operation.

A massive volume of data is sent and received between
the DSP module and the routing module, and a mutual
connection via the backplane inside the same cabinet was
the only method used in the past.
When the above method is used, all modules are put in anelectrically connected state. Therefore, it was notpossible to completely eliminate the risk of a trouble
occurring in a single module affecting the other modules.
On the other hand, using TR-LINK, which can transmit
512-channel audio data in a 32-bit floating point datastate, the data can now be transmitted between modules
using an optical fiber cable. As a result, the DSP moduleand routing module can be installed as completelyseparate units.
The units are completely electrically separated from eachother and thus it is possible to minimize the risk of atrouble occurring in a single unit affecting the entire
system.

Availability and fault tolerance
Hot standby system

Firmware-based system

High-speed startup
The startup time of the entire console system from its
power-off state is approximately 30 seconds.
Even when a critical system error occurs and the entire
system must be restarted, this feature can minimize thesystem downtime.

IO Sharing
Sharing of input audio

to Router

6to Router

Higher integrated circuit with 44-bit high-precision arithmetic
operation capacity

TAMURA has developed a new algorithm that can perform
a 44-bit floating-point arithmetic operation for function
such as an equalizer for which sound quality is particularlyimportant.
The distortion produced by deviation is reduced byincreasing the accuracy of the arithmetic coefficient,making it possible to achieve an unprecedentedly clear
and transparent sound quality.

Studio B

The NT series adopts TAMURA’s own hybrid audio
processing system using the DSP and the FPGA.
The combined use of superior features of both thesedevices significantly improves the arithmetic operation
capacity and provides a higher integrated processor withhigh processing performance for the NT series.
The entire system has been significantly downsized, forexample, a 1U-size DSP unit can perform 256-channel
audio signal processing.
Power consumption has also been considerably reduced
compared with conventional systems because of higher
integrated circuits and a downsized system.

Controls, such as the gain control of a microphone input
shared by multiple systems, are enabled from any system.
Furthermore, the control protect setting is made at anysystem, which enables the gain control to be performed
from a specific system only.
The input audio can be shared among a maximum of eightsystems.

The audio input to a single I/O Frame can be shared
between multiple systems.
For example, you can construct a system that allows amicrophone to be used in each studio from either of thetwo studios.
This feature makes it possible to mutually use the systems
at two studios for emergency backup or use one of thesystems as the Premix Mixer.

Hybrid Audio Processing
Higher integrated processor

The NT series has been built as a firmware-based system
without using general-purpose operating systems such as
Windows and Linux.
Because this system does not require a shutdown
operation, the system can be restarted promptly at anytime.
Furthermore, all operations are always stored in the
backup memory; therefore, the status immediately prior toshutdown is restored when the system is restarted.
Even when the system is involuntarily restarted after a
power trouble or other unexpected accidents, theoperation status will be securely maintained.

The router unit, which is a core component of the system,
has a backup system that is always on hot standby. Thatis, exactly the same unit is in a standby state with thesame operation status as that of the active unit.
The standby system always stores a mirror copy of theactive system’s operation status. Therefore, switching to
the standby system can be performed immediately.
This feature minimizes the system downtime.

Studio A

High-speed data transmission protocol TR-LINK

7

Connection diagram
AUTOMIX function
Sync 1

Sync 2
DSP

Router ( Active )DSP ( Back Up)

Router ( Standby )WindowsLaptop

The AUTOMIX function of NT series models automatessome of the mixing operations.
In broadcast and content production that use several
microphones, an audio mixing engineer must accurately
and immediately control the fader for multiple microphonechannels depending on the situation.
The AUTOMIX function uses network technology toperform an automatic microphone channel fader operation in an effort to lighten the load of the mixing engineer
and provide environment in which the engineer canconcentrate on sound quality adjustment and other tasks.

Conceptual diagram of mixing system environment

Main Specifications of AUTOMIXThe AUTOMIX function of NT series models uses
gainsharing technology to provide the followingfeatures.

Ethernet

MIX

23OUTPUT

Direct soundIndirect soundTR-LINK

Specifications

Specifications
Connectionchannel

SystemDSP CORE

Sampling frequency

48kHz / 96kHz

Supply voltage

Routing cross point

10,240 10,240

Number of signal processing channels

Maximum number of signal processing channels

Word
AES3 / AES3id
Maximum 5 DSP core units(including 1 backup unit)

Number of TR-Link audio channels

512ch

256ch

IO FRAME
Supply voltage
Number of installed slots
IO cards14 slots
8ch Dsub MIC/LINE IN card
8ch BNC AES IN card
HD-SDI card

8ch BNC AES OUT card
AC100-240V 50/60Hz

MADI IO card

Number of TR-Link ports

20 ports

GPIO card

Maximum number of signal processing channels

1,024chSynchronous signal input connector

Specifications
Maximum 4

No. of Automixchannels

16ch

Automix ch format

MonoSample freq

FS 48kConnect ch type

HA/Line Input
GroupM1/M2/M3

Connect ch format

Mono/Stereo/5.1

Connect ch signal path

Depends on theinsertion path

(1) Makes it possible to gain a natural auditorysensation• Produces a sound without the obvious noise gateeﬀect.
• Produces sound right from the start of a speech.
• No need for mixing engineer to bother about levelﬂuctuations.
• Causes no ambience imbalances.
(2) No need to set a threshold level• Ambient noise during a low threshold level does notcause the gate function to activate.
• High threshold levels do not cause the gate to beclosed.
• Even if the threshold level is set in a quiet room, it
will operate properly when there is audience clapping
or a musical performance.
(3) No need to set attack time and hold time

AC100-240V 50/60Hz

8ch Dsub LINE OUT card

ROUTER
Supply voltage

AC100-240V 50/60Hz

1,024chVideo（NTSC/PAL）

Synchronous signal

8Ambientnoise

IO

Item
No. of Automix SHARC DSPs

DSP CORE

1(4) No unnatural muting (no ambient) occurs even
immediately after a speech has ﬁnished and thesubsequent feeling of reverberation is maintained.
(5) The endings of words in a speech are captured
properly.
(6) The quality of ambient noise does not change
when a new speaker starts talking.
(7) No popping noise occurs in the lower frequency
range (caused by gate operation).

BNC connector x 2
X L R connector x 2

9with Tamura Resource Network Technology

with Tamura Resource Network Technology

Excellent operability
Two parameter operation methods
Two methods are available for channel parameter
operation, namely, the center-assign method, whichassigns channels on a panel at a single location, and thechannel-based method, which performs the operation foreach channel as in the case of an analog console.
When you want to concentrate on a single channel sound,the center-assign method is most suitable because it
allows you to operate all parameters at once.
On the other hand, the channel-based method isconvenient when urgency is required, for example, during
live broadcasting, because it allows the engineer tooperate multiple channels at the same time.
These two operation methods are suitable for differentsituations.
NT880 allows operation using either method so that bothmethods can be selected in accordance with the situationand the level of expertise of a mixing engineer.
For the channel-based method in particular, highoperability for a quick response to the situation thatchanges moment by moment is achieved by placing 14
encoders per channel in order to minimize the function
switching operation.

Channel layout editing functions“Add new channels,” “delete channels no longer in use,”or “add a new microphone channel to existing active
channels because another microphone has been added.”
As in the case of these examples, it will be ideal if you canflexibly change the channel layout in accordance with thesituation instead of having a channel layout that is fixed
once it is set.
To enable such an operation, NT880 is provided withsophisticated channel layout editing functions (such as
channel addition, deletion, copying, and cut and insert) on
the touch panel.
This feature intuitively and instantaneously enables mixing
engineers to set up an ideal channel layout.

Flagship model pursuing optimal ease ofoperation to enable high-level creative work

Expandable to large-scale systems
Number of physical faders
NT880 can be configured with up to 150 physical faders.
Two or more consoles in different cabinets can beoperated as a single console system as long as thenumber of installed faders is within the maximum numberof faders. (*1)

Operator-specific section concept
When two or more engineers perform mixing operations at
the same time, other engineers’ work can be interrupted
or their specific settings may be lost if one engineer needs
to perform an operation that affects the entire console.
To prevent such inconveniences, TAMURA has introduced
the section concept.
A ‘section’ specifies the operation range of one engineer.
The extent of the effect from one operation is confined tothe designated section only.
A mixing engineer can also set Pre-Fader Listen (PFL) andAfter-Fader Listen (AFL) solo functions independently foreach section. Therefore, engineers are provided withconditions under which they appear to be working on
separate console systems.
One console system can be divided into a maximum offour sections.

12High-performance processor
NT880 has a control system that is built on theassumption of simultaneous operations by two or moreengineers.
Even if there is a simultaneously imposed workload of fourengineers, there is no delay in the response time of theoperation panel displays or the adjustments made foraudio.
(*1) There are cases in which restrictions apply to the installation position,distance, and other factors.

13with Tamura Resource Network Technology

Specifications

Dimension

ConsoleAudio control parameters

Supply voltage

AC100-240V 50/60Hz

Maximum number of physical faders
Bank / Layer
Number of fader groups

HA Gain150 faders

Trim+24dB~-24dB

6Bank / 2Layer

Delay

5000ms or more

32Group

FilterFilter1 (HPF/Notch)
Filter2 (LPF/Notch)

Audio channel (Fs=48kHz)
Master Bus

Maximum 24 buses(3 surround)Group Bus

Maximum 32 buses

Aux Bus

Maximum 48 buses

N-1 / MT Bus

Maximum 128 buses

AFL+10dBu~-64dBu

Equalizer

4Band(Support for all frequency bands)

Dynamics

Compressor 2 channels
Gate/Expander 1channel

30 Fader1 surround

AFL / PFL

3 stereoPFL

1 stereo

Main Monitor

1 surround+stereo

Sub Monitor

5 channels (Stereo)

40 FaderAudio block diagram
50 Fader
INPUT
IN
KEY
INS

1-1024

MAS
GROUP
AUX
N-1/MT
PFL/AFLDIR OUT

GROUP
EXT
INS

1-32

EXT
TB

14INS
TB

PFL/AFL
1-128(N-1+MT=Max128)

N-1

MTMAS
AUX
PFL/AFL
1-128(N-1+MT=Max128)

MTMAS

MAS

PFL/AFL

N-1
EXT
TB

MAS
GROUP
AUX
N-1/MT
PFL/AFLDIR OUT

EXT
TB

1/2/3(SURR)
AUX

AUX

1-48

MONITOR
1(SURR)
2-6(ST)MONI

60 FaderPFL

15