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    G.711, G.721, G.726 and G.728 Codecs Voip

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    This document discusses various audio compression algorithms or codecs that are commonly used for voice over internet protocol (VoIP) applications. It describes codecs such as G.711, G.722, G.723, G.726, G.727, G.728, and G.729, providing information on their bitrates and applications. It also discusses VOCAL Technologies' software libraries that implement these codecs and are optimized for digital signal processors.

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    G.711, G.721, G.726 and G.728 Codecs Voip

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    aaes2
    219 views5 pages
    This document discusses various audio compression algorithms or codecs that are commonly used for voice over internet protocol (VoIP) applications. It describes codecs such as G.711, G.722, G.723, G.726, G.727, G.728, and G.729, providing information on their bitrates and applications. It also discusses VOCAL Technologies' software libraries that implement these codecs and are optimized for digital signal processors.

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    VoIP Codecs 2

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    VoIP Codecs

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    This document discusses various audio compression algorithms or codecs that are commonly used for voice over internet protocol (VoIP) applications. It describes codecs such as G.711, G.722, G.723, G.726, G.727, G.728, and G.729, providing information on their bitrates and applications. It also discusses VOCAL Technologies' software libraries that implement these codecs and are optimized for digital signal processors.

    Copyright:

    © All Rights Reserved
    Download as PDF, TXT or read online from Scribd
    Download as pdf or txt
    219 views5 pages

    G.711, G.721, G.726 and G.728 Codecs Voip

    Uploaded by

    aaes2
    This document discusses various audio compression algorithms or codecs that are commonly used for voice over internet protocol (VoIP) applications. It describes codecs such as G.711, G.722, G.723, G.726, G.727, G.728, and G.729, providing information on their bitrates and applications. It also discusses VOCAL Technologies' software libraries that implement these codecs and are optimized for digital signal processors.

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    The document discusses different audio compression algorithms used in VoIP including G.711, G.721, G.726, G.728, G.723 and G.729. It also discusses features and resource requirements of these algorithms.

    Some of the audio compression algorithms discussed include G.711, G.721, G.726, G.728, G.723 and G.729. G.711 is a 64 kbps algorithm, G.721 is a 32 kbps ADPCM algorithm, G.726 extends G.721 to include rates of 40, 24 and 16 kbps. G.728 is a 16 kbps CELP algorithm.

    G.729 is an 8 kbps CS-ACELP algorithm approved by ITU-T. It processes signals with 10 ms frames and has a 5 ms look-ahead for a total delay of 15 ms. Features include full and half duplex modes, passing ITU test vectors, and a common compressed frame interface to support multiple coders.

    Codecs VoIP

    VOCAL Technologies, Ltd. Home page

    Codecs VoIP

    VOCAL Technologies, Ltd. Home page | Contact us | About us

    G.711 | G.722.1 | G.722.2 G.723.1 | G.726


    G.727 | G.728 G.729/G.729 Annex A | G.729 Annex B
    G.729 Annex D | G.729 Annex E GSM EFR | GSM FR
    GSM HR

    Codecs VoIP Software

    VOCAL offers a wide options of audio compression algorithms, mainly from the ITU (International
    Telecommunications Union) and the ETSI, (European Telecommunicaton Standard Institute)

    G.711, G.721, G.726 and G.728 Codecs VoIP


    The CCITT approved the G.711 recommendation on Pulse-Coded Modulation (PCM) •-Law or A-Law
    in 1984. It is a 64 kbps compressed stream and is a common reference for speech compression quality.
    G.711 encoding/decoding is usually done within codecs although a technique to convert linear samples
    to PCM (A-Law or •-Law) is straightforward.

    G.721 is a 32 kbps Adaptive Differential Pulse Code Modulation (ADPCM) speech compression
    algorithm. It produces toll quality speech. Under error free transmission, G.721 has a slightly worse
    perceived quality than G.711. The quality of G.721 significantly deteriorates when several such links are
    used in tandem. With transmission error rates higher than 10.4, the perceived quality of G.721 is better
    than G.711.

    G.726 extends the G.721 ADPCM to include 40, 24 and 16 kbps, as well as 32 kbps. G.726 at 40 kbps
    performs comparable to G.711.

    G.728 Low-Delay Code Excited Linear Prediction (LD-CELP) compression is a 16 kbps compression.

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    Codecs VoIP

    This has an algorithmic coding delay of 0.625 ms. Compared with G.721, G.728 tends to score worse in
    objective, but better in subjective testes. A characteristic of CELP algorithms is that they tend to perform
    poorer than ADPCM in the presence of background noise.

    Features

    - Full or half duplex modes of operation.


    - Pass ITU test vectors.
    - Common compressed speech frame stream interface to support systems with multiple speech coders
    (G.723, G.729, et al.).
    - Optimized for high performance on leading edge DSP architectures.
    - Multi-tasking environment compatible.

    Configurations

    - DAA interface using linear codec at 8.0 kHz sample rate.


    - Direct interface to 8.0 kHz PCM data stream (A-law or -law).
    - North American/International Telephony (including caller ID) support available.
    - Simultaneous DTMF detector operation available - (less than 150 hits on Bellcore test tape typical).
    - MF tone detectors, general purpose programmable tone detectors/generators available.
    - Line echo cancellation (G.165 compliant) available.
    - Where multiple speech coders (G.723, G.729 et al.) are available, coder selection can occur at run-
    time.
    - Data/Facsimile/Voice Distinction available.
    - Complete facsimile systems available - modulations (V.34fax, V.17, et al.) and protocols (T.30) as a
    facsimile terminal or facsimile relay configuration.
    - Complete data modem systems available - modulations (V.90, V.34, et al.) and protocols (V.42, PPP
    framing, et al.).
    - Data/Facsimile/Voice command sets available.
    - Various startup procedures available (V.8 and V.8bis).
    - Multiple ports can be executed on a single DSP.

    Example Resource Requirements (ADSP-21xx)

    - G.728 at 16 kbps requires 30 MIPS


    - G.726/G.721 encoder requires 3.5 MIPS, decoder requires 3.7
    - A popular non-standard configuration of G.726/G.721 encoder requires 3.2 MIPS, decoder requires 3.0
    MIPS

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    Codecs VoIP

    G.723 Codecs VoIP


    VOCAL Technologies modem/telephony software libraries include the G.723 dual rate speech coder
    optimized for execution on DSP architectures from leading silicon suppliers. This software is modular
    and can be executed as a single task under a variety of operating systems or it can executed standalone
    with its own microkernel. The G.723 speech coder recommendation was developed for use in
    multimedia platforms, in particular those specified by the H.32x series recommendations. It provides
    two compressed stream bit rates, 5 1/3k bps and 6.4k bps. The higher bit rate is of greater quality.

    The coder processes signals with 30 ms frames and has a 7.5 ms look-ahead. Relative to the G.729/
    G.729A coders, the G.723 speech coders pass DTMF tones through with less distortion.

    Both the 5 1/3k bps and 6.4k bps rates are mandatory for the encoder and decoder. A G.723 frame
    stream may switch between the two rates at any 30 ms frame boundary.

    The line interface may be an analog front end (codec and DAA) or a digital interface such as T1/E1,
    switched 56 and ISDN. The upper end of this software offers a direct binary and speech frame interface.

    Features

    - Full and half duplex modes of operation.


    - Passes ITU test vectors.
    - Common compressed speech frame stream interface to support systems with multiple speech coders
    (G.729, G.728, G.726 et al).
    - Optimized for high performance on leading edge DSP architectures.
    - Multi-tasking environment compatible.

    Configurations

    - DAA interface using linear codec at 8.0 kHz sample rate.


    - Direct interface to 8.0 kHz PCM data stream (A-law or -law).
    - North American/International Telephony (including caller ID) support available.
    - Simultaneous DTMF detector operation available - (less than 150 hits on Bellcore test tape typical).
    - MF tone detectors, general purpose programmable tone detectors/generators available.
    - Line echo cancellation (G.165 compliant) available.
    - Where multiple speech coders (G.729, G.728, G.726 et al.) are available, coder selection can occur at
    run-time.
    - Data/Facsimile/Voice Distinction available.
    - Complete facsimile systems available - modulations (V.34fax, V.17, et al.) and protocols (T.30) as a
    facsimile terminal or facsimile relay configuration.
    - Complete data modem systems available - modulations (V.90, V.34, et al.) and protocols (V.42, PPP

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    Codecs VoIP

    framing, et al.).
    - Data/Facsimile/Voice command sets available.
    - Various startup procedures available (V.8 and V.8bis).
    - Multiple ports can be executed on a single DSP.

    Example Resource Requirements (ADSP-2181)

    - Encoder 5 1/3k bps requires 18 MIPS


    - Encoder 6.4k bps requires 26 MIPS
    - Decoder (5 1/3k bps or 6.4k bps) requires 2 MIPS

    G.729/G.729 Annex A Codecs VoIP


    VOCAL Technologies modem/telephony software libraries include G.729 and G.729 Annex A speech
    coders optimized for execution on DSP architecture from leading silicon suppliers. This software is
    modular and can be executed as a single task under a variety of operating systems or it can be executed
    standalone with its own microkernel.

    G.729 is an 8 kbps Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP) speech


    compression algorithm approved buy ITU-T. G.729 Annex A is a reduced complexity version of the
    G.729 coder.

    G.729 Annex A speech coder was developed for use in multimedia simultaneous voice and data
    applications like DSVD. The coder processes signals with 10 ms frames and has a 5 ms look-ahead
    which results in a total of 15 ms algorithmic delay. The input/output of this algorithm is 16 bit linear
    PCM samples that are converted from/to 8 kbps compressed data stream.

    The line interface may be an analog front end (codec and DAA) or a digital interface such as T1/E1,
    switched 56 and ISDN. The upper end of this software offers a direct binary and speech frame interface.

    Features

    - Full and half duplex modes of operation.


    - Passes ITU test vectors.
    - Common compressed speech frame stream interface to support systems with multiple speech coders
    (G.723, G.728, G.726 et al).
    - Optimized for high performance on leading edge DSP architectures.
    - Multi-tasking environment compatible.

    Configurations

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    Codecs VoIP

    -DAA interface using linear codec at 8.0 kHz sample rate.


    -Direct interface to 8.0 kHz PCM data stream (A-law or •-Law).
    -North American/International Telephony (including caller ID) support available.
    -Simultaneous DTMF detector operation available - (less than 150 hits on Bellcore test tape typical).
    -MF tone detectors, general purpose programmable tone detectors/generators available.
    -Line echo cancellation (G.165 compliant) available.
    -Where multiple speech coders (G.723, G.728, G.726 et al.) are available, coder selection can occur at
    run-time.
    -Data/Facsimile/Voice Distinction available.
    -Complete facsimile systems available - modulations (V.34fax, V.17, et al.) and protocols (T.30) as a
    facsimile terminal or facsimile relay configuration.
    -Complete data modem systems available - modulations (V.90, V.34, et al.) and protocols (V.42, PPP
    framing, et al.).
    -Data/Facsimile/Voice command sets available.
    -Various startup procedures available (V.8 and V.8bis).
    -Multiple ports can be executed on a single DSP.

    Example Resource Requirements (ADSP-2181)

    -G.729 Annex A Encoder requires 8.7 MIPS, 6762 words of PM, and 1512 words of DM
    -G.729 Annex A Decoder requires 2.1 MIPS, 3950 words of PM, and 725 words of DM
    -G.729 Annex A Encoder/Decoder requires 10.8 MIPS, 7720 words of PM, and 1918 words of DM

    VOCAL Technologies, Ltd. modem software libraries include a complete range of ETSI / ITU / IEEE
    compliant modulations, optimized for execution on ANSI C and leading DSP architectures (ADI, AMD,
    ARM, CEVA, LSI Logic ZSP, MIPS and TI). This software is modular and can be executed as a single
    task under a variety of operating systems or it can execute standalone with its own kernel.

    VOCAL is an active contributor to the International Telecommunications Union (ITU),


    Telecommunication Industry Association (TIA), ANSI T1E1.4, the Institute of Electrical and
    Electronics Engineers (IEEE) and the European Telecommunication Standards Institute (ETSI).

    VOCAL Technologies, Ltd. Home page | Contact us | About us


    G.711 | G.722.1 | G.722.2
    G.723.1 | G.726 | G.727
    G.728 | G.729/G.729 Annex A | G.729 Annex B
    G.729 Annex D | G.729 Annex E | GSM EFR
    GSM FR | GSM HR

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