Voice Codec G.729

Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an.

Feb 2, 2006 - This document explains voice codec bandwidth calculations and. For example, a G.729 voice payload size of 20 ms two 10 ms codec.

G.729AB codec uses VAD, CNG and DTX to reduce bandwidth. It is described in ITU-T G.729 Annex A and Annex B. In addition, it also uses the G.729 Annex B specifications to reduce transmission during periods of silence.

G.729AB codec uses VAD, CNG and DTX to reduce bandwidth.

G.729AB codecWideband audio codecMobile, VoIP and Voice ConferencingReal-time multi-channel implementationOptimized for DSPs, RiSC, CISC processorsITU-T G.729 Annex A and B compliant

G.729AB codec is suitable as an internet wideband audio codec for VoIP applications, where normal conversation characteristics may be exploited in order to reduce bandwidth usage.Contact us to discuss your G.729AB vocoder application requirements.

VOCAL s G.729AB source code is optimized for performance on leading DSPs and RISC/CISC processors from TI, ADI, AMD, ARM, Intel and other vendors. VOCAL offers G.729AB voice compression as a standalone algorithm, as part of a modular software library suite, as well as with a VoIP stack. Custom designs are also available to meet your specific G.729AB  codec needs.G.729AB

G.729AB is a reduced complexity Conjugate-Structure Algebraic-Code-Excited Linear Prediction CS-ACELP speech compression algorithm that uses discontinuous transmission to reduce bandwidth. It is described in ITU-T G.729 Annex A and Annex B.

G.729AB requires 10 ms input frames and generates frames of 80 bits in length. Since G.729AB is based on the Code-Excited Linear Prediction CELP  model, each 80 bit frame produced contains linear prediction coefficients, excitation code book indices, and gain parameters that are used by the decoder in order to reproduce speech. The inputs/outputs of G.729AB are 16 bit linear PCM samples that are converted from/to an 8 kbps compressed data stream. G.729AB has the same total algorithmic delay of 15 ms as the G.729 speech coder.

G.729AB contains the same reduced complexity modifications present in the G.729A speech coder. In addition, it also uses the G.729 Annex B specifications to reduce transmission during periods of silence. In conversations, it is normal that periods of silence may occur with no voice present in the signal. During these times, it is possible to calculate the background noise characteristics and transmit these parameters instead, and only update values if the background noise changes significantly.

The G.729AB encoder uses Voice Activity Detection VAD to determine if voice is present in the input signal. If so, the frame is constructed in accordance with G.729A. Otherwise, the noise characteristics are calculated and a Silence Insertion Description SID frame is sent instead; only 10 bits are present in a SID frame. The decoder will use this information for Comfort Noise Generation CNG, as talkers expect some level of background noise during periods of silence. It is only necessary to transmit new SID frames when the background noise level changes. Discontinuous Transmission DTX reduces bandwidth usage by only transmitting voice or SID frames; during silence, only SID frames are sent when necessary to describe the current level of background noise.FeaturesCompliant with G.729, Annex A, Annex B specificationsFull and half duplex modes of operationPasses ITU test vectorsOptimized for high performance on leading edge DSP architecturesMultichannel implementationMulti-tasking environment compatibleConfigurationsDAA interface using linear codec at 8.0 kHz sample rateDirect interface to 8.0 kHz PCM data stream A-law or μ-law North American/International Telephony including caller ID support availableSimultaneous DTMF detector operation available – less than 10 talkoff hits on Bellcore test tape set MF tone detectors, general purpose programmable tone detectors/generators availableData/Facsimile/Voice Distinction availableCommon compressed speech frame stream interface to support systems with multiple speech codersDynamic speech coders selection if multiple speech codecs availableCan be integrated with Acoustic Echo Canceller, G.168 Line Echo Canceller and Tone Detection/Regeneration modulesAvailable with VoIP stackMultiple ports can be executed on a single DSPPlatforms

VOCAL s speech coders are available for the following platforms. Please contact us for specific G.729AB platforms.ProcessorsOperating SystemsTexas Instruments – C6000 TMS320C62x, TMS320C64x, TMS320C645x, TMS320C66x, TMS320C67x, DaVinci, OMAP, C5000 TMS320C54x, TMS320C55x Analog Devices – Blackfin, ADSP-21xx, TigerSHARC, SHARCPowerPCMIPS – MIPS32, MIPS64, MIPS4KcARM – ARM7, ARM9, ARM9E, ARM10E, ARM11, StrongARM, ARM Cortex-A8, Cortex-M1Intel / AMD – x86, x64 both 32 and 64 bit modes Google AndroidApple iOS / iPhone / iPad MacOSUnix,  Linux, μClinux, BSDMicrosoft Windows ACM / RTC / CE / MobileSymbianeCOS / eCOSProWind River VxWorksGreen Hills IntegrityVOCAL LANsENDMicrium μCOSMore Information.

G.729 is an audio data compression algorithm for voice that compresses digital voice in and high-quality audio cannot be transported reliably with this codec. G.729 Annexes -  G.729 Annex A -  G.729 Annex B -  Other extensionsG 729 Codec Asterisk Software Digiumwww.digium.com Products Asterisk SoftwareCachedDigium.

G.729

G.729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding CS-ACELP. 1

Because of its low bandwidth requirements, G.729 is mostly used in Voice over Internet Protocol VoIP applications where bandwidth must be conserved, such as conference calls. Standard G.729 operates at a bit rate of 8 kbit/s, but there are extensions, which provide rates of 6.4 kbit/s Annex D, F, H, I, C and 11.8 kbit/s Annex E, G, H, I, C for worse and better speech quality, respectively.

G.729 has been extended with various features, commonly designated as G.729a and G.729b. The differences between these versions are noteworthy because they each have additional properties and thus different use-cases:

G729: original codec. Uses high-complexity algorithm

G729A or A annex: medium complexity variant of G.729 and it is compatible with G729. It is less complex but has slightly lower voice quality

G729B or B annex: G729 with silence suppression and not compatible with the previous ones.

G729AB: G729A with silence suppression and only compatible with G729B.

Dual-tone multi-frequency signaling DTMF, fax transmissions, and high-quality audio cannot be transported reliably with this codec. DTMF requires the use of the RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals as specified in RFC 2833.

Embedded variable bit rate, wideband

G.729a is a compatible extension of G.729, but requires less computational power. This lower complexity, however, bears the cost of marginally reduced speech quality.

G.729a was developed by a consortium of organizations: France Telecom, Mitsubishi Electric Corporation, Nippon Telegraph and Telephone Corporation NTT

Sampling frequency 8 kHz/16-bit 80 samples for 10 ms frames

Fixed bit rate 8 kbit/s 10 ms frames

Fixed frame size 10 bytes for 10 ms frame

Algorithmic delay is 15 ms per frame, with 5 ms look-ahead delay

G.729a is a hybrid speech coder which uses Algebraic Code Excited Linear Prediction ACELP

The complexity of the algorithm is rated at 15, using a relative scale where G.711 is 1 and G.723.1 is 25.

PSQM testing under ideal conditions yields Mean Opinion Scores of 4.04 for G.729a, compared to 4.45 for G.711 μ-law

PSQM testing under network stress yields Mean Opinion Scores of 3.51 for G.729a, compared to 4.13 for G.711 μ-law

Some VoIP phones incorrectly use the description G729a/8000 in SDP e.g. this affects some Cisco and Linksys phones. This is incorrect as G729a is an alternative method of encoding the audio, but still generates data decodable by either G729 or G729a - i.e. there is no difference in terms of codec negotiation. Since the SDP RFC allows static payload types to be overridden by the textual rtpmap description this can cause problems calling from these phones to endpoints adherring to the RFC unless the codec is renamed in their settings since they will not recognise G729a as G729 without a specific workaround in place for the bug.

G.729 has been extended in Annex B G.729b which provides a silence compression method that enables a voice activity detection VAD module. It is used to detect voice activity in the signal. It also includes a discontinuous transmission DTX module which decides on updating the background noise parameters for non speech noisy frames. It uses 2-byte Silence Insertion Descriptor SID frames transmitted to initiate comfort noise generation CNG. If transmission is stopped, and the link goes quiet because of no speech, the receiving side might assume that the link has been cut. By inserting comfort noise, analog hiss is simulated digitally during silence to assure the receiver that the link is active and operational.

Recently, G.729 has been extended with Annex J to provide support for wideband speech and audio coding, i.e., the transmitted acoustic frequency range is extended to 50 Hz - 7 kHz. The respective extension to G.729 is referred to as G.729.1. The G.729.1 codec is hierarchically organized: Its bit rate and the obtained quality are adjustable by simple bitstream truncation.

G.729 includes patents from several companies and is licensed by Sipro Lab Telecom. Sipro Lab Telecom is the authorized Intellectual Property Licensing Administrator for G.729 technology and patent pool. 3 4 5 6 In a number of countries, the use of G.729 may require a license fee and/or royalty fee. 5

AIM IP, a non-practicing entity based in Mission Viejo, CA has made complaints for infringement of U.S. Patent No. 5,920,853 to a number of companies. 7 8 The patent contains lookup tables including the sequence 1486, 2168, 3751 which is integral to g.729. 9

The European counterpart of this patent, according to Google patent records appears to have been withdrawn: 10

G.729 is assigned the static payload type 18 for RTP by IANA. 11 The correct rtpmap description for this payload type is G729/8000.

Both G.729a and G.729b use the same rtpmap description as G.729. G.729a and G.729b is indicated using annexb no or annexb yes respectively. G.729 Annex B G.729b is the default in absence of annexb parameter in Session Description Protocol. 12

Comparison of audio coding formats

International Telecommunications Union, Standardization Sector ITU-T, Study Group 15 1993-1996, Recommendation G.729, March 1996.

ITU-T January 2007. G.729 : Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction CS-ACELP PDF : i. Retrieved 2009-07-21.

VoiceAge Corporation 2007-10-14. G.729 Licensing. Archived from the original on 2007-10-14. Retrieved 2009-09-17.

a b Sipro Lab Telecom 2007-10-25. FAQ G.729 and G.723.1. Archived from the original on 2007-10-25. Retrieved 2009-09-17.

Sipro Lab Telecom 2006-10-29. G.729 IPR Pool. Archived from the original on 2006-10-29. Retrieved 2009-09-17.

Real-Time Transport Protocol RTP Parameters. Iana.org. Retrieved 2013-09-18.

S. Casner, P. Hoschka July 2003. MIME Type Registration of RTP Payload Formats. Retrieved 2013-02-27.

ITU-T Recommendation G.729 - technical specification

G.729 Error Recovery for Internet Telephony

Sipro Lab Telecom administers the patent pools for G.723.1 and G.729

Voiceage s G.729 implementation

Retrieved from https://en.wikipedia.org/w/index.php.title G.729 oldid 661077277

Categories: Audio codecsSpeech codecsITU-T recommendations.

G.729 digitizes analog voice signals producing output at 8 kilobits per second Kbps with 8:1 Microsoft s Real-Time Codec RTC for VoIP optimization.

Digium s implementation of the G.729 codec allows Asterisk software to convert audio between G.729 and any other supported codec. Many IP telephones and.