Voice Compression Index
Wideband and Narrowband speech (audio) codecs (vocoder)
Voice Codecs analyze speech and convert it into digital form. On the transmit side of the connection speech signals are transformed/compressed into electronically transmitted digital information (encode). On the receive side of the connection, a codec with matching capabilities accepts the digital signal and recreates an approximation of the original analog signal (decode).
Adaptive Digital's voice codecs - click on codec below or scroll down for product page link and description.G.711 WAVEFORM CODER
G.711, G.711 Appendix1 & Appendix2- 
Availability C54x, C55x, C6xx, & ARM9
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.
G.722
G.722 - Audio Coder Availability C54x, C55x, C6xx
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.
G.722 C55x™DSP supports optional feature on the decoder side: Packet Loss Concealment. This feature is implemented using a proprietary technique developed by Adaptive Digital.
G.722.1
G.722.1 - Audio
Coder Availability C54x, C6xx
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.
G.722.2 ADAPTIVE MULIT-RATE WIDEBAND CODEC
G.722.2 - WB Audio
Coder Availability C55x, C6xx
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
G.723.1 VOICE CODER
G.723.1 - Voice
Coder Availability C54x, C55x, C6xx
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.
G.726 VOICE CODER
G.726 - Voice Coder Availability C54x, C55x, C6xx
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.
G.728 VOICE CODER
G.728 - Voice Coder Availability C54x, C55x, C6xx
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.
G.729 VOICE CODER APPENDICES A, B, AB, and D
G.729,
G.729A, G.729B, G.729AB, G.729D Availability C54x, C55x, C6xx, ARM9
G.729A available for reduced MIPS requirements
G.729B available for voice activity detection, silence frame insertion, and comfort noise generation
G.729D supports multiple bit rates
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).
GSM ADAPTIVE MULTI-RATE VOICE CODER
GSM
AMR Availability C54x, C55x, C6xx
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.
GSM FULL RATE SPEECH CODER
ADT
GSM FR Availability C64x
Preliminary - C54x, C55x
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.
LPC DEFENSE COMMUNICTION VOICE CODER
LPC - Voice Coder
Availability C67x
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.
SELECTABLE MODE VOCODER
SMV - Voice
Audio Coder Availability C64x
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 SPEECH CODEC
EVRC Availability C64x
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.
EVRC-B
EVRC - Enhanced Variable Rate Codec B Availability C64x
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.
MIXED EXCITED LINEAR PREDICTIVE
MELP Availability C55x, C6xx
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.
ENHANCED MIXED EXCITED LINEAR PREDICTIVE
MELPe - enhanced Mixed Excited Linear Predictive Availability C64x
Adaptive Digital's enhanced-MELP (MELPe) a low bit rate vocoder supporting 2400, and 1200 bps operates at half the rate of the MELP standard. Enhancements include: Improved encoding/decoding, transcoding between 2400, 1200 and 600 bps bit streams, and noise preprocessing for removing background noise.
ADAPTIVE DIGITAL 4800 PROPRIETARY VOICE CODEC
ADT4800 - Voice Coder Availability C54x
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.
ADAPTIVE DIGITAL 9600 PROPRIETARY VOICE CODEC
ADT9600 - Voice Coder Availability C54x
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.
LOW BIT RATE VOCODER
iLBC Availability C54x
, C55x , C64x , C64x+
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.
