PLYMOUTH MEETING, Pa., July 1, 2013 /PRNewswire/ -- Adaptive Digital Technologies (Adaptive Digital) today announced complete, ready-to-use Linux- and Android-based voice over IP (VoIP) solutions for Sitara™ AM335x ARM® Cortex™- A8 processors Texas Instruments Incorporated (TI): LnxVoice™ for Linux, and AnVoice™ for Android. By choosing Adaptive Digital's VoIP Solutions for Sitara processors, innovators can create feature-rich, low-power VoIP applications.
- Complete, ready-to-use, VoIP solution, including High-Definition Acoustic Echo Cancellation and Wideband codecs, and supporting SIP and peer-to-peer applications.
- Supports BeagleBone and BeagleBone Black open-source computers, which can run either Linux or Android.
- Simple API: Developers can easily integrate Adaptive Digital's VoIP solution for TI's Sitara AM335x ARM Cortex- A8 processors into Linux, and Android-based applications.
Adaptive Digital's VoIP Engine™/SIP Reference Kits, include both a voice over internet protocol engine software development kit (SDK) and a Session Initiation Protocol (SIP) SDK, which can be used together to accelerate the development of cutting-edge VoIP applications while delivering the best end-user experience.
VoIP Engine SDK features high-definition acoustic echo cancellation, noise reduction and automatic gain control (AGC) for voice quality enhancement (VQE), as well as speech compression (G.711, G.729AB, and G.722 and other optional codecs), RTP and Jitter Buffering.
Adaptive Digital's SIP SDK provides a customizable solution to quickly add SIP-based dial and receive phone call features into software applications. It supports: registration, call initiation, call-acceptance and call-teardown for VoIP telephones.
Reference Design Kits for Sitara include:
- SIP Phone Sample Project with source code
- VoIP Engine SDK (includes evaluation version library, header file and docs)
- SIP SDK (includes evaluation version library, header file and docs)
- SDK Quick Start Guide
- Developer Quick Start Guide
The included sample SIP Phone app is a fully functioning SIP phone. The app can be configured to connect to a standard SIP server. The sample SIP Phone app is also configured to place outgoing phone calls as well as receive inbound phone calls. Furthermore, Adaptive Digital's VoIP Engine/SIP Reference Kit supports peer-to-peer VoIP for applications that do not require SIP, opening the door to many new applications.
The VoIP Engine/SIP Reference Kit API is clean and simple to use. In fact, developing a VoIP-enabled app is as easy as customizing one of the sample SIP Phone apps, which are provided by Adaptive Digital in source code format as part of the SDK.
Included in the Reference Kit is G.722, a wideband compression algorithm, passing 7 kHz of audio bandwidth rather than the 3.5 kHz that is carried by wired phones and cell phones. The result is crystal clear voice, the likes of which is impossible in the public switched telephone network.
"The combination of Adaptive Digital's Voice Engine technology and TI's high-performance, low-power Sitara AM335x ARM processors enables customers to develop best-in-market, full-featured VoIP applications," said Rogerio Almeida, marketing manager, Sitara ARM processors, TI. "With Adaptive Digital's development kit, customers can accelerate time to market while still delivering the features and voice quality that matter to end users."
SIP Phone Application
- SIP Client Protocol
- RTP/Jitter Buffer
- Voice Conferencing (up to 4 users)
- G.711 mu-law and a-law with packet loss concealment
- G.729AB 8 kbps speech compression
- G.722 16 kbps speech compression
- Noise Reduction
- HD Acoustic Echo Cancellation*(Wideband) - *Operates on most Android handsets without customization/tuning.
- Automatic Gain Control
- Tone Generation
- Tone Relay Transmit
- Peer-to-Peer Operation
- Fully Configurable via GUI
- Tone transmit
- Tone receive
- CSS transmit
- CSS receive
- Acoustic Delay Measurement
AMR-NB and AMR-WB
"Adaptive Digital and TI have a long history of producing robust voice applications for traditional telephony and VoIP markets," said Brian McCarthy, president, Adaptive Digital Technologies. "TI's high performance, low-power Sitara AM335x ARM Cortex A-8 processors, along with Adaptive Digital's LnxVoice and AnVoice Reference Kits, enable Sitara customers to develop custom VoIP desktop, intercom, and mobile device applications, while vastly reducing time-to-market cycles."
Let Adaptive Digital make voice work for you. If you have custom functionality and features in mind, but don't know quite where to start, Adaptive Digital can customize a solution to fit your specific requirements.
BeagleBone or BeagleBone Black open-source computers featuring Texas Instrument's Sitara AM335x ARM Cortex-A8 processor
BeagleBone Audio Cape: Provides stereo audio input and output for the BeagleBone hardware platform.
AnVoice VoIP Engine/SIP Demo Application is currently available for free Download via Adaptive Digital's extranet. Registration required.
For more information regarding LnxVoice and AnVoice Reference kit running on TI's Sitara AM1x ARM9™ processor, or Adaptive Digital's VoIP Solutions on other TI platforms, contact Adaptive Digital, 610-825-0182, Sales: Ext 120.
About VoIP Engine
VoIP Engine is a framework that bundles together many algorithms such as conferencing, vocoders, noise reduction, echo cancellation, etc. required in a VoIP application. Although VoIP Engine is more integrated than an algorithm, it is still not an application. It is a data processing engine. VoIP Engine has no interface to drivers or peripherals and performs processing solely at the request of the host application. This makes VoIP engine portable for use in conjunction with any application or operating system.
VoIP Engine is more than just a collection of algorithms packaged in a library. It connects them together. For example, in a PCM to Packet configuration, VoIP Engine is fed a PCM stream by an application. It processes the PCM through echo cancellation, noise reduction, AGC, speech compression, RTP, etc. and it returns a complete RTP packet to the application. Similarly, in the opposite direction, the application feeds VoIP Engine with an RTP packet and VoIP Engine returns PCM samples.
VoIP Engine is intended use is in VoIP enabled handsets or desktop phones. Although VoIP Engine is not tied to any particular software environment, it was designed with Android™, iPhone™ Windows, and Linux in mind.
About Adaptive Digital Technologies
Adaptive Digital Technologies is a leading global provider of fully optimized echo cancellation, voice compression, voice quality, audio & video algorithms, and both DSP chips & turnkey solutions for both IP and traditional telecommunications systems / applications, and VoIP for mobile digital devices (including both Android [http://www.adaptivedigital.com/product/anVoip-anVoice.htm , and iPhone [http://www.adaptivedigital.com/product/iPVoice-VoIP-voice-engine.htm] architecture).
Adaptive Digital's solutions support a low-cost product development model with short time-to-market. For more information concerning Adaptive Digital Technologies, visit http://www.adaptivedigital.com or contact sales at 610-825-0182 x120
About the Texas Instruments Design Network
Adaptive Digital Technologies is a member of the TI Design Network, a premier group of independent, well-established companies that offer products and system-level design and manufacturing services complementing TI's semiconductors to a worldwide customer base to accelerate product innovation and time-to-market. Network members provide product design, hardware and software system integration, turnkey product design, RF and processor system modules, reference platforms, software development, proof-of-concept design, feasibility studies, research, certification compliance, prototyping, manufacturing, and product life cycle management. For more information about the TI Design Network, please visit http://www.ti.com/designnetwork.
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SOURCE Adaptive Digital Technologies