IntegrIT CrystalSpeech  is the latest voice processing technology in the real-time speech processing software enabling hands-free full-duplex communications. CrystalSpeech is ideally suited for conferencing terminals, smart phones, communicators, videophones, dispatcher boards, speech synthesis and recognition systems, VoIP solutions, etc. Speech enhancement technology includes intellectual echo and noise suppressors providing natural speech quality that selectively recognizes active speaker even in noisy places. This allows conversations in a wide range of conditions with extremely high echo level and environmental noise. Echo cancellation technology adapted for use in mobile devices such as notebooks, communicators and gadgets where audio quality is limited due to the mechanical resonances, small speakers and high level of microphone-speaker acoustic feedback.

Features:

• adaptive echo cancellation providing comfort conversations even in presence of high acoustic echo (up to 500 msec)
• nonlinearity compensator suppressing remaining echo caused by nonlinear distortions in acoustic path or mechanical resonances
• smart noise suppression system reducing the level of electrical and environmental noise with minimum speech distortions
• voice-activated automatic gain control system balancing speech level without noise increase in the pauses
• antihowling filters preventing distortions due to the high level of acoustic coupling
• the set of user profiles for easy parameter selection (car, street, office, etc.)
• multiplatform implementation for compatibility (ARM9E, ARM11, Cortex A8, TI TMS320C6xxx, Windows, Linux)

Applications:

• conferencing terminals, web VoIP terminals
• smartphones, communicators, gadgets
• hands-free car kits
• videophones, dispatcher boards, speech synthesis and recognition systems (ARS), hands-free sets
• VoIP equipment
  

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell ARMADA
  
• x86 (Windows, Linux)

NatureDSP VoIP Engine – the core component of multichannel telephony and acoustic IP equipment in particular IP telephones, PBX, VoIP gateways, dispatching consoles and similar equipment providing clear voice function over packet channels.

The software configurable NatureDSP VoIP Engine can be customized for various applications, processors and can work in soft real time environment (Windows, Linux) as well as in hard real time environment to minimize delays and provide regular service.

List of supported codecs allows interfacing with well recognized IP equipment and soft phones. Adaptive jitter buffering together with advanced packet loss concealment allow keeping excellent voice quality even in hard conditions with reduced network bandwidth, packets reordering, floating network delays etc. Advanced speech enhancement algorithms, in particular for linear (LEC) and acoustic (AEC) echo canceling and noise suppressing, voice AGC and other provide robust predictable behavior in wide range of channel conditions. Acoustic echo canceler allows to use NatureDSP VoIP Engine in microphone/speaker systems in full duplex hands-free mode.

Availability:

The NatureDSP VoIP Engine is available in binary and 'C' source codes form for the following platforms:

• Texas Instruments TMS320C64xx, DaVinci (DSP-BIOS)
• Marvell Sheeva/KirKwood/Dove, ARM9E, ARM11 (Linux)
• PC: Windows/Linux
• Other platforms and OS are under request

:

• adaptive echo cancellation providing comfort conversations even in presence of high line echo (up to 128 msec)
• nonlinearity compensator suppressing remaining echo caused by nonlinear distortions in electrical or acoustic path or mechanical resonances
• antihowling filters preventing distortions due to the high level of acoustic coupling

Supported plarforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove
• Tensilica
  
• Microsoft Windows
• Linux

• smart noise suppression reducing the level of electrical and environmental noise with minimum speech distortions
• the set of user profiles for easy parameter selection (car, street, office, etc.)
• low resource (MIPS/memory) consumption allows to run on almost all CPUs
• multiplatform implementation for compatibility with multiple operation systems
• antihowling filters preventing distortions due to the high level of acoustic coupling

• hands-free sets and car kits
• conferencing terminals
• smartphones, communicators, gadgets
• speech recorders
•  videophones
• VoIP equipment

• Texas Instruments C64xx, DaVinci, OMAP 35xx
• ARM9e, ARM11, Cortex A8, etc.
• Marvell Kirkwood, Sheeva, ARMADA
• Intel x86

is a royalty-free codec for Voice over IP (VoIP) network. IntegrIT iLBC delivers speech quality better than G.729A and equal to G.729E, while offering substantially better quality over congested networks with packet loss. It is designed for narrowband speech and results in a payload bit rate of 13.33 kbps for 30 ms frames and 15.20 kbps for 20 ms frames. The codec enables graceful speech quality degradation in the case of lost frames, which occurs in connection with lost or delayed IP packets

Features:

• Operates at 13.3 / 15.2 kbps bitrate
• Frame size 30 ms for 13.3kbps, and 20 ms for 15.2kbps
• Voice quality exceeds G.729A and G.723.1
• High robustness to packet loss
• Low delay and high packet loss robustness for low-bit rate codecs
• Start state encoding
• Pitch enhancement
• Packet loss concealment
• Demo available for target and PC

Applications

• VoIP
• Telephony

List of supported platforms:
     • Texas Instruments C64xx, DaVinci
     • MS Windows

iLBC is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

Triple rate (6.4/8/11.8 kbit/sec) CS_ACELP vocoder conforming complete ITU-T G.729 recommendation. It can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products.

Features:

• Coding rates 8 kbps (G.729 main body), 6.4 kbps (Annex D) or 11.8 kbps (Annex E)
• Reduced complexity version G.729AB for increased performance
• Integrated voice activity detector, comfort noise generator
• Music detection (for 11.8 kbps)
• Sampling rate 8 kHz
• 16-bit linear signal input
• C-callable program interface
• Multi-channel capable
• Capable of in-band synchronization
• The encoder and decoder meet all ITU G.729I compliance and interoperability requirements
• Demo available for target and PC
• Fully portable ANSI C code

Applications:
     • Communication Devices
     • VoIP
     • Telephony

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove, ARMADA
  
• Microsoft Windows
• Linux

IntegrIT G.729 is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

 Low-delay code excited linear prediction (LD-CELP) voice codec conforming complete ITU-T G.728 recommendation. It provides coding of speech signals at 16 kbit/s and at reduced rates 9.6 and 12.8 kbit/s. Integrated packet loss concealment synthesizes the speech during the periods when the bit stream is missed or errored making the signal erasures inaudible. Voice codec can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products.

Features:

• Coding rates 16 kbps (G.728 main body), 9.6 kbps or 12.8 kbps (Annex H)
• Packet loss concealment (Annex I)
• Sampling rate 8 kHz
• 16-bit linear signal input
• C-callable program interface
• Multi-channel capable
• The encoder and decoder meet all ITU G.728 compliance and interoperability requirements.
• Demo available for target and PC

Applications

• Communication Devices:
• VoIP
• Telephony
  

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• MS Windows

IntegrIT G.728 is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

16/24/32/40 kbit/sec ADPCM voice codec ITU-T G.726 recommendation. It can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products.

Features:

• Coding rates 16, 24, 32 and 40 kbps
• A-law, mu-law and 14-bit PCM interfaces
• Sampling rate 8 kHz
• C-callable program interface
• Multi-channel capable
• The encoder and decoder meet all ITU G.726 compliance and interoperability requirements.
• Demo available for target and PC
• Applications

Communication Devices:

• VoIP
• Telephony
• Digital storage

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove, Marvell ARMADA
• x86, (Microsoft Windows, Linux)

G.726 is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

Dual rate wideband (50…7000 Hz) voice codec conforming the ITU-T G.722.1 recommendation. It can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products

Features:

• Coding rate 24 or 32 kbps
• Frame rate 20 msec
• Sampling rate 16 kHz
• Very low CPU usage
• C-callable program interface
• Multi-channel capable
• The encoder and decoder meet all ITU G.722.1 compliance and interoperability requirements
• Demo available for target and PC
• Applications

Communication Devices

• VoIP
• Telephony
• List of supported platforms:

Supported targets:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove
• Microsoft Windows
• Linux

G.722.1 is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

Dual rate vocoder conforms to ITU-T G.723.1 recommendation.
It can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products

Features:
     • Coding rate 5.3 or 6.3 kbps
     • Sampling rate 8 kHz
     • 16-bit linear signal input
     • C-callable program interface
     • Multi-channel capable
     • Capable of in-band synchronization
     • The encoder and decoder meet all ITU G.723.1 compliance and interoperability requirements
     • Demo available for target and PC
     • Fully portable ANSI C code
     • Applications

Communication Devices
     • VoIP
     • Telephony

List of supported platforms:
     • Texas Instruments C64xx, DaVinci
     • ARM9E, ARMv5, ARMv7
     • Marvell Sheeva/KirKwood, Marvell Dove
     • Microsoft Windows
     • Linux

G.723 is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. 

 

64 kbit/sec A/u-law voice codec fully compliant with ITU-T G.711 recommendation. Packet loss concealment (PLC) feature enables it for use in a wide range of internet/mobile applications such as VoIP and videoconferencing products. The objective of PLC is to generate a synthetic speech signal to cover missing data (erasures) in a received bit stream. It tries to synthesize the signal that has the same timbre and spectral characteristics as the missing signal. If the erasures are not too long, and the erasure does not land in a region where the signal is rapidly changing, the erasures may be inaudible after concealment. VAD/CNG feature provides means to use of discontinuous transmission in a packet-based communication system that can significantly reduce the transmission rate and hence improve the bandwidth efficiency. It uses generic payload format and may also be used with other speech codecs without built-in DTX capability such as G.726, G.727, G.728, and G.722. The VAD algorithm makes a voice activity decision based on multiple parameters such as the full band energy, the low band energy, the zero-crossing rate and a spectral measure. This provides robust decision over a wide range of conditions and the level of ambient noise.

Features

• Coding rate 64 kbps
• A-law, u-law encoding
• Sampling rate 8 kHz
• PLC compliant with Appendix I of G.711
• VAD/CNG compliant with Appendix II of G.711 and interoperable with G.729B
• C-callable program interface
• Multi-channel capable
• Demo available for target and PC
• Fully portable ANSI C code

Applications

• VoIP
• Telephony

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove, Marvell ARMADA
  
• x86 (Microsoft Windows, Linux)

G.711PLC is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods.

Is a standard ACELP vocoder adapted by the 3rd Generation Partnership Project (3GPP). It is an Adaptive Multi Rate-Narrow Band (AMR-NB) speech codec. This vocoder is used mainly in 3rd generation mobile telephony devices to compress toll-quality speech at 8000 samples/second. GSM-AMR codec has eight basic bit rates, 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 Kbit/s

Features

• Eight coding rates in range of 4.75 to 12.2 kbps
• Sampling rate 8 kHz
• Full 3GPP TS 26.073 compliance
• 16-bit linear signal input
• C-callable program interface
• Multi-channel capable
• Demo available for target and PC
• Fully portable ANSI C code

Applications

• Communication Devices
• VoIP
• Telephony

List of supported platforms:

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove
• Microsoft Windows
• Linux

GSM AMR-NB is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. Evaluation version for for PC and your target is available upon request. 

is Adaptive Multi Rate Wide Band vocoder conforming the ITU-T G.722.2 recommendation. Codec provides excellent speech quality in a clean environment at low bitrates. Higher bitrates are useful in noisy conditions and in the presence of music. It can be used in a wide range of applications such as multimedia devices, visual telephony, wireless telephony, and videoconferencing products.

• Nine coding rates in range of 6.60 to 23.85 kbps
• Sampling rate 16 kHz
• ITU G.722.2 compliance and interoperability requirements.
• 16-bit linear signal input
• C-callable program interface
• Multi-channel capable
• Demo available for target and PC
• Fully portable ANSI C code

• Communication Devices
• VoIP
• Telephony

• Texas Instruments C64xx, DaVinci
• ARM9E, ARMv5, ARMv7
• Marvell Sheeva/KirKwood, Marvell Dove
• Microsoft Windows
• Linux

GSM AMR-WB is delivered with fully automated IntegrIT Testing Environment (ITE) for target platform based on reference ITU-T vectors set along with extended IntegrIT proprietary vectors and methods. Evaluation version for for PC and your target is available upon request.