Speex Codec

Speex codec supports customized high definition (HD) voice over IP (VoIP) and file-based compression applications.  Speex yields good quality speech using Code Excited Linear Prediction (CELP) encoding techniques, is available with multiple bit-rates, and is robust to lost or delayed packets. Contact us to discuss your Speex voice codec application requirements.

VOCAL’s Speex codec source code is optimized for performance on DSP and conventional processors from TI, ADI, ARM, Intel and other leading vendors. Our Speex vocoder software is available for licensing standalone, as part of a modular software library, or with a VoIP stack for different integration options with your application. Custom designs are also available for our clients to meet your specific needs.

Speex

• 3 different sampling rates: 8 kHz, 16 kHz, and 32 kHz (narrowband, wideband, and ultra-wideband)
• Speex encoding is controlled most of the time by a quality parameter that ranges from 0 to 10
• Vary encoder complexity dynamically by adjusting how lookup is performed
• Dynamically adjust its bit-rate to adapt to the complexity of the audio being encoded
• Voice activity detection (VAD) with comfort noise generation (CNG) to reproduce background noise
• Discontinuous transmission (DTX) mode to transmit only when speech is present or comfort noise parameters must be updated
• Perceptual enhancement by decoder which enhances sound quality subjectively (although not objectively)
• Algorithmic delay is 30ms in narrowband mode, 34ms in wideband mode

Features

• Internet wideband audio codec
• Patent and royalty-free software
• Optimized for high performance on leading DSP architectures
• Integration of narrowband and wideband using an embedded bit-stream
• Wide range of bit-rates available (from 2 kbps to 44 kbps)
• Standard bit rates include 2.15, 3.95, 5.95, 8, 11, 15, 18.2 and 24.6 kbps
• Dynamic bit-rate switching and Variable Bit-Rate (VBR)
• VAD integrated with VBR
• Variable complexity
• Ultra-wideband mode at 32 kHz (up to 48 kHz)
• Intensity stereo encoding option
• Code Excited Linear Prediction (CELP) based
• Multichannel implementation
• 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 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 VoIP stack