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G.711 | G.722 | G.722.1 | G.722.2 |G.723.1 | G.726 | G.728 | G.729 | GSM AMR | GSM FR | LPC | SMV | EVRC | EVRC B | MELP | MELPe | iLBC
The Adaptive Digital G.711 waveform coder is an implementation of the ITU G.711 PCM standard. Adaptive Digital’s G.711 coder converts between 8-bit mu-law or a-law PCM and 16 bit uniformly quantized PCM. Although G.711 conversion is built into the DSP’s serial port, it is often necessary to perform G.711 conversion in software to retain flexibility.
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The Adaptive Digital Technologies G.722 Audio Coder is a real-time implementation of the ITU G.722 audio coder. It is used with many applications that require audio frequency bandwidth coding such as video conferencing, multimedia, and speaker/microphone digital telephony.
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The Adaptive Digital Technologies G.722.1 Audio Coder is a real-time implementation of the ITU G.722.1 standard. Developed to deliver wideband speech at lower bit rates than G.722. The G.722.1 audio coder encodes 16 kHz sampled audio signals for transmission over 24 and 32 kbps channels, and provides 7 kHz of audio bandwidth.
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The Adaptive Digital Technologies G.722.2 Audio Coder is a real-time implementation of the ITU G.722.2 audio coder also referred to as the Adaptive Multi-Rate Wideband (AMR-WB) codec. Not only does AMR-WB provide advanced voice quality over the existing narrowband standards, but it is also very robust against transmission errors due to multi-rate operation and adaptation.
Coding Rate: 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 and 23.85 kbps
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The Adaptive Digital Technologies’ G.723 voice coder is a real-time implementation of the ITU G.723.1 voice coder. It is used with many applications that require high quality, robust speech reproduction. G.723.1 is specified in numerous Voice-Over-Packet environments such as Voice-Over-IP, and Voice-Over-ATM.
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The Adaptive Digital Technologies G.726 voice coder is used in many applications
that require high quality, robust speech reproduction. G.726 provides rates of 16, 24, 32, or 40 kbps. Applications include video conferencing systems, multimedia, flight recording, ISDN, and satellite communications. ADT offers both a high performance (Low MIPS) and low memory version.
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The Adaptive Digital Technologies’ G.728 voice coder is a real-time implementation of the ITU G.728 toll quality voice coder. ITU G.728 has rapidly gained acceptance for many applications including satellite, cellular, and video conferencing systems.
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G.729A available for reduced MIPS requirements
G.729B available for voice activity detection, silence frame insertion, and comfort noise
generation
The Adaptive Digital Technologies G.729 voice coder software is a real-time implementation of the ITU G.729 voice coder. ITU G.729 is used with many applications that require toll quality robust speech reproduction. The software runs on the Texas Instruments (TI) TMS320C54X family of Digital Signal Processors (DSPs).
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GSM-AMR is an adaptive multi-rate speech coder that has been standardized for use in Third Generation (3G) mobile telephony. The coder supports eight bit rates: 12.2, 10.2, 7.95, 7.40, 6.7, 5.9, 5.15, and 4.75 kbps. The coder uses algebraic code excited linear prediction (ACELP) as the compression method. AMR also includes Voice Activity Detection (VAD) and Discontinuous Transmission (DTX) as an added way to save bandwidth by sending fewer bits per second when the user is not speaking. GSM-AMR was developed to maintain high speech quality under a wide range of transmission conditions.
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The Adaptive Digital GSM (Global System for Mobile) FR vocoder is a real-time implementation of GSM 06.10 RPELTP (Regular Pulse Excitation-Long Term-Prediction-Linear Predictive Coder) vocoder. GSM FR was the first digital speech-coding standard used in GSM digital mobile phone system.
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The Adaptive Digital Technologies LPC-C67X voice coder is a real-time implementation of one of the LPC-10 vocoder. LPC-10 is one of the early vocoder standards and has been adopted and used in numerous defense communication systems.
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The Adaptive Digital Technologies Selectable Mode Vocoder (SMV) is a real-time implementation of the 3GPP2 standardized vocoder. The vocoder, originally developed for cellular telephony, also has other applications in VoIP. The SMV codec encodes 8 kHz sampled voice audio signals for transmission, and compresses the signals to rates of 8.55 4.0, or 2.0 kbps.
--------------------------------------------------------Enhanced Variable Rate CODEC (EVRC) is a speech codec used in CDMA networks based on relaxed code-excited linear predictive (RCELP) algorithm. The development of EVRC in 1995 offered the mobile carriers more capacity on their networks while not increasing the amount of bandwidth or wireless spectrum needed. Given that EVRC operates at a low bit-rate while delivering high quality speech. It is ideal for variable rate operations and robust CDMA networks.
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The significant enhancement in EVRC-B is the use of 1/4-rate frames that were not used in EVRC. The EVRC-B makes use of the intermediate coding rates through increased awareness of the nature of the individual speech samples. Voice over IP (VoIP) applications operating over low bandwidth dial-up and wireless networks require such enhancements for efficient use of the bandwidth.
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Adaptive Digital's MELP (Mixed-Excitation Linear Predictive)
vocoder is a real-time implementation of the Federal Standard 2400 bps speech
coder, suitable for use by OEM customers for VoIP, telecom, military and
government, and other low bit-rate compressed speech applications.
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Adaptive Digital's MELPe (Mixed-Excitation Linear Predictive) vocoder, known as military standard MIL-STD-3005 and NATO STANAG 4591, is a low rate coder that operates at 2400 bps. MELPe is suitable for use by OEM customers for VoIP, telecom, military and government, and other low bit-rate compressed speech applications.
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The Adaptive Digital Technologies ADT-4800 voice coder software is a real-time
implementation of Adaptive Digital's proprietary 4800 bps speech compression
algorithm. This algorithm provides good speech quality at a low bit rate.
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The Adaptive Digital Technologies ADT-9600 voice coder software is a real-time
implementation of Adaptive Digital's proprietary 4800 bps speech compression
algorithm. This algorithm provides good speech quality at a low bit rate.
Available as standard or low MIPS.
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iLBC is a royalty-free codec for Voice over IP (VoIP) networks. iLBC delivers speech quality better than G.729A and equal to G.729E, while offering significantly better quality over congested networks with packet loss.
SITE LINKS
HOMEVOCODERS
ADT 4800 ADT 9600 G.711 G.722 G.722.1 G.722.2 AMR-WB G.723.1 G.726 G.728 G.729, G.729 AB GSM AMR GSM FR LPC EVRC EVRC-B MELP MELPe SMV iLBCRELATED LINKS
G.168 NEC & LEC G.168 Lite G.168 EC CHIP 'C6424 G.168 EC CHIP 'C641x ACOUSTIC EC ACOUSTIC EC G2 LC IPP REFERENCE DESIGN DSP RESOURCE WIZARD