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G.722.1 Codec

G.722.1 vocoder

G.722.1 is used for hands-free operation in HD VoIP systems with low frame loss

G.722.1 Vocoder

  • Wideband audio codec (50 to 7,000 Hz)
  • High quality speech for HD VoIP
  • Real-time multi-channel implementation
  • Optimized for DSPs, RiSC, CISC processors
  • ITU G.722.1 compliant

VOCAL’s G.722.1 codec is used for hands-free operation in HD VoIP systems with low frame loss as well as as an internet wideband audio codec for VoIP applications. Contact us to discuss your G.722.1 codec application requirements.

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VOCAL’s G.722.1 software is optimized for leading DSPs and RISC/CISC processors from TI, ADI, AMD, ARM, Intel and other vendors. G.722.1 vocoder may be licensed as a standalone algorithm, as a software library suite, as well as with a VoIP stack for different application integration options. Custom designs are also available.


The G.722.1 algorithm describes a wideband speech coder that provides an audio bandwidth of 50 Hz to 7 kHz, operating at a bit rate of 24 kbps or 32 kbps for hands-free operation in HD VoIP systems with low frame loss or as an internet wideband audio codec for VoIP applications.

The G.722.1 voice compression algorithm is based on transform technology, using a Modulated Lapped Transform (MLT). It operates on 20 ms frames (320 samples) of audio. Because the transform window (basis function length) is 640 samples and a 50 per cent (320 samples) overlap is used between frames, the effective look-ahead buffer size is 20 ms. Hence the total algorithmic delay of 40 ms is the sum of the frame size plus look-ahead. All other delays are due to computational and network transmission delays.

The digital input to the G.722.1 encoder may be 14, 15 or 16 bit 2’s complement format at a sample rate of 16 kHz. It is handled in the same way as outlined in ITU Recommendation G.722. The analogue and digital interface circuitry at the encoder input and decoder output conforms to the specifications described for G.722.


  • DAA interface using linear codec at 8.0 kHz sample rate
  • Direct interface to 8.0 kHz PCM data stream (A-law or μ-law)
  • North American/International Telephony (including caller ID) support available
  • Simultaneous DTMF detector operation available – (less than 10 talkoff hits on Bellcore test tape set)
  • MF tone detectors, general purpose programmable tone detectors/generators available
  • Data/Facsimile/Voice Distinction available
  • Common compressed speech frame stream interface to support systems with multiple speech coders
  • Dynamic speech coders selection if multiple speech codecs available
  • Can be integrated with Acoustic Echo Canceller, G.168 Line Echo Canceller and Tone Detection/Regeneration modules
  • Available with integrated VoIP stack


  • Full and half duplex modes of operation
  • Passes ITU test vectors
  • Compliant with G.722.1 specification
  • Optimized for high performance on leading edge DSP architectures
  • Multichannel implementation
  • Multi-tasking environment compatible



VOCAL’s optimized vocoder software is available for the following platforms. Please contact us for specific G.722.1 codec supported platforms.

Processors Operating Systems
  • Texas Instruments – C6000 (TMS320C62x, TMS320C64x, TMS320C645x, TMS320C66x, TMS320C67x), DaVinci, OMAP, C5000 (TMS320C54x, TMS320C55x)
  • Analog Devices – Blackfin, ADSP-21xx, TigerSHARC, SHARC
  • PowerPC
  • MIPS – MIPS32, MIPS64, MIPS4Kc
  • ARM – ARM7, ARM9, ARM9E, ARM10E, ARM11, StrongARM, ARM Cortex-A8, Cortex-M1
  • Intel / AMD – x86, x64 (both 32 and 64 bit modes)
  • Google Android
  • Apple iOS / iPhone / iPad & MacOS
  • Unix,  Linux, μClinux, BSD
  • Microsoft Windows ACM / RTC / CE / Mobile
  • Symbian
  • eCOS / eCOSPro
  • Wind River VxWorks
  • Green Hills Integrity
  • Micrium μCOS

More Information

VOCAL Technologies, Ltd.
520 Lee Entrance, Suite 202
Amherst New York 14228
Phone: +1-716-688-4675
Fax: +1-716-639-0713