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Audio Codecs
Speech Codecs
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Audio Processing Subsystem
Speech Codecs

The following is the list of Speech Decoders and Encoders implemented on ARC processors.  As always, our team has the capabilities to quickly port these solutions on other popular DSP and RISC processors.

G.711 Codec
G.711 is an ITU-T standard for speech codecs that provides toll quality speech at 64kbps using either A-law or mu-law PCM methods. While mu-law is generally used in North America and Japan, A-law is used in Europe and the rest of the world. This falls in the category of waveform speech coders which try to compress the waveform as it sample by sample.

The Incube G.711 codec is very compact and efficient implementation. Data Sheet

G.726 Codec
G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32 and 40 kbps. These four bit-rates are often referred to by the bit size of a sample, which are 2 bits, 3 bits, 4 bits and 5 bits respectively, since the sampling frequency is fixed to 8 kHz. This codec falls in the category of waveform speech codecs. Data Sheet
G.729 AB Codec
G.729 is an ITU-T speech codec that adopts Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) scheme. It is a parametric speech codec that operates at 8 kbps and finds applications in telephony over packet networks and Voice over IP. G.729A is the reduced complexity version of G.729, which marginally sacrifices the speech quality to achieve reduction in complexity. The annex B of G.729 is a silence compression scheme which uses a VAD module to detect voice activity in the signal, a DTX module which decides on updating the background noice parameters for non-speech frames, and a CNG module that generates the required comfort noise akin to the old analog hiss.

Incube G.729 codec is an implementation of the low-complexity version of G.729 with Annex A and B. Data Sheet

AMR-NB Codec
AMR-NB or simply AMR is a patented audio data compression scheme optimized for speech coding. AMR uses the Algebraic Code Excited Linear Prediction (ACELP) in addition to DTX, VAD and CNG to reduce bandwidth usage during silence periods. It uses “link adaptation” to select one of eight different bit-rates based on link conditions. It was adopted as the standard speech codec by 3GPP and is widely used in GSM and UMTS. Data Sheet
AMR-WB Codec
Adaptive Multi Rate – WideBand is a parametric speech codec that provides excellent speech quality due to the use of wider speech bandwidth of 50-7000 Hz as compared to narrow band codecs that which are optimized for POTS wireline quality of 300-3400 Hz. AMR-WB uses the Algebraic Code Excited Linear Prediction (ACELP). The sampling frequency used is 16 kHz as opposed to 8 kHz used for narrow-band codecs. AMR-WB is codified as G.722.2, an ITU-T standard speech codec.

Incube AMR-WB codec has been heavily optimized to enable its use even in low-power applications. Data Sheet

AMR-WB+ Codec
AMR-WB+ is an extension of AMR-WB. The main improvement is the use of transform coding additionally to ACELP. This greatly improves the performance of this codec for audio signals. Automatic switching between transform coding and ACELP provides both good speech and audio quality with moderate bit-rates. It adds support to stereo signals, higher sampling rates and a wide range of bit-rates. While doing all this, AMR-WB+ manages to be backward compatible to AMR-WB.

Building on its highly efficient AMR-WB codec implementation and expertise in audio codec development, Incube AMR-WB+ codec implementation provides a very attractive MHz proposition for the end customer using AMR-WB+ codec. Data Sheet

EVRC-B Codec
Enhanced Variable Rate Codec-B (EVRC-B) is a 3GPP2 standard 4GV codec for cellular phones in CDMA2000 systems. EVRC-B is an extended version of EVRC which is a Relaxed Code Excited Linear Prediction (RCELP) based speech coding algorithm. In addition to RCELP, EVRC-B uses Prototype Pitch Period (PPP) approach for coding of stationary voice frames and Noise Excitation Linear Prediction (NELP) for efficient coding of unvoiced or noise frames. Thus it provides lower data rates compared to EVRC, as it includes 1/4 rate frames, null frames and erasure frames compressing each frame of data into one of the four bit rate frames of 9.6, 4.8, 2.4 and 1.2kpbs. EVRC-B has operation points (0,1 and 2) facilitating superior flexibility in rate assignment supporting average bit rate in the range of 9.3 - 4.8 kbps.
Incube's EVRC-B codec, with all the above features, is efficiently optimized for low MHz and low power real-time operation.
Speex Codec
Speex is an open source lossy speech codec for mainly VOIP applications. Incube's Speex codec is a highly optimized codec supporting narrowband, wideband and ultra- wideband (for 8, 16 and 32kHz sampling rates respectively). It supports both variable bit rate mode (VBR) and average bit rate mode (ABR) operations resulting in a very low bit rate ranging between 2 – 44 kbs. Its features include voice activity detection (VAD), comfort noise generation (CMG) and discontinuous transmission (DTX).

Data Sheet

SILK Codec
SILK is CELP based speech algorithm used for packet based voice communications. Silk provides scalability in several dimensions to target a range of operating environments. The features of SILK codec make it useful in different network conditions are control of bit rate, packet rate, resilience to packet loss and use of discontinuous transmission (DTX). SILK codec can be set at different complexity levels to suite different processor capabilities. SILK decoder integrated with sample rate converter, it converts output signal to 8, 12, 16 or 24 KHz frequency based on the user option. Though SILK’s main applications are for internet based voice communications, its super-wideband capabilities (24 kH) make it a potential candidate for internet based audio transmission.

Incube's implementation of SILK codec supports all the above features and is highly optimized for low MHz and smaller memory footprint.
Speech Processing:
Acoustic Echo Canceller (AEC):
Acoustic Echo Canceller (AEC) is a technique used to suppress acoustic echo in teleconferencing and hands-free telephony applications. Incube's AEC consists of efficient adaptive filter to provide over 40dB echo suppression up to echo tail-lengths of 64 ms. It operates in full-duplex mode with a robust double talk detector with high rate of double talk detection and negligible false alarms. It operates in both narrowband and wideband modes (for sampling frequencies of 8 and 16 kHz respectively). The Incube's AEC software provides a complete solution with features such as noise suppression, voice activity detection (VAD), comfort noise generation (CMG) and automatic gain control (AGC).

Data Sheet

Automatic Gain Control (AGC)
Incube's Automatic Gain Controller (AGC) is well suited for telephonic applications with smooth gain variations and well calibrated thresholds for noise detection (to avoid unnecessary amplification of noise during no-speech periods) and maximum gain (to avoid saturation of speech).

Data Sheet

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