ADAPTIVE DIGITAL = VOICE QUALITY
THERE WILL ALWAYS BE A NEED FOR VOICE
Providing Crystal Clear Communication for all Voice Applications
Adaptive Digital Technologies – A Driving Technology for Leading Brands Company:
Proud provider of Third Party Software Solutions & Design Integration Services for the following providers
Arm Community
MELPe
MELPe is a US DoD and NATO secure voice codec that supports rates of 600 bps, 1200 bps, and 2400 bps Defense Applications have included: Secure VoIP, Aerospace/Airline Communication Systems, Low Bandwidth Radio Communications, Ground Forces Communications.
Use Case MELPe: Military Defense Low-Bandwith Secure Voice
Portable Communication: Used in specialized ruggedized field devicesSecure Terminal Equipment (STE): Utilized in government secure phone systemsSecure Radio Systems: Used in Software Defined Radios (SDR) for secure voice communication
Problem: Application requirements include low-bandwidth, secure, and reliable voice communications in challenging environments. Particularly in military, satellite, and tactical radio systems where data rates are very low (600-2400 bps) and packet loss/channel errors are common, requiring robust speech compression and error resilience.
MELPe software is capable of running multi streams (multi-channel) together, either encoding and decoding concurrently.
The MELPe codec supports three different vocoder bitrates: 2400, 1200, and 600 bps. The basic 2400 bps bitrate vocoder uses a 22.5 ms frame of speech consisting of 180 8000-Hz, 16-bit speech samples. The 1200 and 600 bps bitrate vocoders each use three and four 22.5 ms frames of speech, respectively.
These reduced bitrate vocoders internally use multiple 2400 bps parameter sets with further processing to strategically remove redundancy. The payload sizes for each of the bitrates are 54, 81, and 54 bits for the 2400, 1200, and 600 bps frames, respectively.
The MELPe algorithm distinguishes between voiced and unvoiced speech and encodes each differently. Unvoiced speech can be coded with fewer information bits for the same quality. MELPe codec includes enhanced noise reduction for challenging environments.
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Key Use Cases & Applications:
Secure Voice & Radio: Provides secure, low-rate speech for tactical radios, satellite comms, and STE (Secure Terminal Equipment).
Military & Defense: Primary use is in secure tactical communications, soldier radios (JTRS, SRW), and interoperability between different platforms. MELPe is designed to perform well in challenging environments.
Satellite Communications: Compresses voice for efficient transmission over limited bandwidth satellite links.
Software Defined Radios (SDR): Integrated into SDR platforms (like JTRS) for interoperability and efficient voice over IP (VoIP).
Mobile & VoIP: Used in mobile apps (Android, iOS) and VoIP systems for secure, bandwidth-efficient communication, often with reference implementations for testing.
VoIP (Voice over IP): Enables secure, low-bandwidth voice calls on networks where quality is crucial and bandwidth is limited.
Key Features Enabling Use Cases:
Low Variable Bit Rates: Supports low data rates (1200, 600 bps) and can dynamically switch rates.
Noise Reduction: Enhanced noise preprocessing (NPP) improves speech clarity in noisy environments.
Scalability: Able to adapt its data rate to channel conditions.
Standardization: Adheres to U.S DoD and NATO STANAG-4591, ensuring interoperability.
High Definition AEC
Enterprise‑Grade Acoustic Echo Cancellation.
Full-Duplex High Definition Acoustic Echo Cancellation algorithm. HD AEC eliminates audible echo in VoIP applications.
Use Case HD AEC: Railway Intercom and Cabin Intercom Product Family
Train Engineer / Conductor - Communication from the train operator or conductor to the passengers Cabin Communication - This intercom allows passengers in cabins to communicate with the staff
Problem: When a speaker is physically close to a microphone—especially in devices like intercoms, conferencing systems, or phones—several acoustic problems appear: strong near-field coupling, high sound pressure level at the mic, and feedback/echo, where the microphone picks up audio coming from the device’s own loudspeaker.
An Acoustic Echo Canceller (AEC) helps by predicting and removing the loudspeaker’s contribution from the microphone signal, even when the physical setup is sub-optimal.
Here’s how it helps:
It models what the loudspeaker is outputting
The AEC receives a digital copy of the audio being played through the loudspeaker. It uses this as the reference signal.
It estimates how sound travels from the speaker to the microphone
This includes: distance between speaker and mic reflections in the room resonances of the device’s enclosure distortions from the loudspeaker
AEC uses an adaptive filter (usually an NLMS or Kalman-type filter) to learn this “acoustic path.”
It subtracts that estimated echo from the microphone input
Because the loudspeaker audio is known, the AEC subtracts its filtered version from the mic signal.
If done well, the mic signal only contains the user’s voice, not the device’s own output.
Why this is especially helpful when the speaker is very close to the mic
When the loudspeaker is near the mic:
Echo becomes stronger
The mic picks up the loudspeaker output with high amplitude. AEC has more signal to subtract, but the model is clearer.
Direct path dominates reflections
Because the speaker is close, the AEC mostly has to model a simple, short impulse response. This can actually make adaptation faster.
Prevents feedback / self-excitation
AEC removes strong near-field speaker energy before it recirculates in the audio system.
VoIP Engine: Embedded VoIP Software That Accelerates Time to Market
VoIP Engine™ eliminates years of VoIP algorithm development by delivering a field proven VoIP media engine that OEMs and solution providers can integrate quickly and reliably. Focus on product differentiation—while relying on a trusted, production ready voice stack.
Bidirectional VoIP Notification & Intercom Software
Adaptive Digital delivers a robust, bidirectional, true full‑duplex VoIP software platform designed for next‑generation doorbell, call‑box, intercom, IoT, and IP notification systems.
Use Case IP Intercom and Notification Software: Reliable, Clear Voice Communication for Clinical Environments
Nurse Call System: Hands Free Voice Communication Between Patients and Care Staff
Problem: In modern healthcare environments, rapid and clear communication between patients and nursing staff is essential. The solution must prioritize patient safety, and staff efficiency while providing clear reliable communication with hands free operation between patient and staff.
Adaptive Digital’s IP/SIP‑based Intercom and Notification Software enables a next‑generation Nurse Call System that provides real‑time, hands‑free, bidirectional voice communication between patient rooms, nurse stations, and central control rooms.
The solution improves response times, patient satisfaction, and staff efficiency while integrating seamlessly into existing hospital IP infrastructure.
IP Intercom Voice Engine Capabilities for Nurse Call Environments
Adaptive Digital’s Voice Engine is optimized for real‑world hospital conditions:
- HD Acoustic Echo Cancellation (HD‑AEC)
Clear speakerphone performance in patient rooms, and reflective sterile surfaces - Noise Reduction
Suppresses alarms, HVAC, carts, and background conversation - Automatic Gain Control (AGC)
Maintains consistent volume regardless of patient position - Packet Loss Concealment & Jitter Buffering
Reliable audio on shared hospital networks - Voice Activity Detection / Comfort Noise Generation (VAD/CNG)
- Tone Generation & Tone Relay – For nurse call integration and alert signaling
- Packet Loss Concealment – Maintains audio quality during network congestion
- Auto Level Controls
- Voice Codecs
- G.711 (broad compatibility)
- G.722 (HD voice for clarity)
- G.729AB (bandwidth-efficient deployments)
- Optional Voice Capabilities (As Needed)
Available to meet specific hospital infrastructure or compliance requirements:
- G.723.1A
- G.726
- G.728
- Tone Detection – For advanced signaling and nurse call workflows
Healthcare‑Focused Features
- High‑Definition Acoustic Echo Cancellation (HD‑AEC)
Enables natural, two‑way hands‑free conversations in patient rooms, operating theaters, and nurse stations—even under reflective surfaces and open acoustics. - Excellent Voice Quality for Clinical Clarity
Ensures intelligible speech for patient requests, staff instructions, and emergency communications where misunderstandings are not acceptable. - Hands‑Free Operation
Ideal for infection‑controlled environments such as:- Patient rooms
- ICUs
- Operating theaters
- Isolation wards
- Push‑to‑Talk Option
Available for clinical workflows requiring controlled or half‑duplex communication, such as staff paging or restricted access areas.
Patient‑to‑Nurse Communication Flow
- Patient Initiates a Call
A patient presses a bedside nurse call button or touch interface. - Instant IP Voice Connection
The system establishes a VoIP call from the patient room to the assigned:- Nurse station
- Mobile nurse terminal
- Central monitoring station
- Hands‑Free, Full‑Duplex Conversation
Nurses and patients communicate simultaneously using true full‑duplex audio, enabling natural conversation without push‑to‑talk delays.
Why Adaptive Digital for Nurse Call Systems?
- Proven full‑duplex VoIP audio quality
- Designed for clinical and safety‑critical use
- Standards‑based SIP and RTP/SRTP support
- Customizable to existing nurse call platforms
- Scalable from single wards to entire hospital campuses
Clear Audio in Clinical Environments
Advanced voice algorithms filter background noise from medical equipment, carts, and alarms, ensuring intelligible speech.
Leading Global Provider of Voice Algorithms and VoIP Solutions across a wide variety of platforms.
Adaptive Digital expertise dramatically improves the quality and clarity of your speech communication application
Leveraging Existing and Emerging Technologies
Adaptive Digital delivers fielded, scalable, state-of-the-art voice enhancement algorithms/solutions, flexible configuration options, and real-world experience enabling exceptional voice call performance across each users’ environment.
Our Expertise
VOICE CLARITY
Voice Quality is as much a science as it is an art; it is not a one design fits all solution. Adaptive Digital can dramatically improve the quality and clarity of your speech communication application.
LEADING GLOBAL PROVIDER
Adaptive Digital is the leading global provider of voice algorithms and VQE solutions across many platforms. Our real-world experience enables exceptional voice call performance across each users’ environment.
TECHNICAL SUPPORT
At Adaptive Digital Technologies, our technical support is second to none.
Your Product Guarantee:
OUR PRODUCT AND SERVICE OFFERINGS DRAW FROM EXTENSIVE INDUSTRY EXPERIENCE.
All Adaptive Digital standard ITU, ETSi, and GSM products come with a compliancy guarantee, to assure product performance on your application platform/device. Your Adaptive Digital custom products come with a pre-defined performance guarantee which assures your satisfaction in your products performance.