VoIP Vocoders
There are several considerations when evaluating a particular voice codec for an application.
Each codec has a set of unique characteristics which may affect their suitability for a particular function and/or platform.
Many of today's voice applications will obtain greater clarity through the use of wideband voice coders. The traditional PSTN limitations no
longer need to be applied to modern voice communication systems. Wideband voice coders operate at higher sample rates such as 16 and
32 KHz. This offers the opportunity for greater clarity and improved latency (using the proper voice coders and packet handling).
Combined with a wideband speakerphone, the user is offered a vivid shift in expectations, perhaps comparable to the switch from standard
TV to high definition TV (HDTV). This high definition voice processing, when incorporated into a VoIP system, will drive user expectations
in many HD VoIP systems that our customers will be developing.
VOCAL's embedded software libraries include a complete range of ETSI / ITU / IEEE compliant algorithms,
in addition to many other standard and proprietary algorithms. Our software is optimized for execution on ANSI C
and leading DSP and RISC architectures (TI, ADI, AMD, Intel, ARM, MIPS, ARC, CEVA, LSI Logic ZSP, etc.).
Current resource requirements for each platform are being updated. Please contact us directly for specific
information. Reference MIPS/memory requirements for certain processors are available for comparison.
Tables of many VoIP vocoders with their associated PSQM/PSQM+ values under various network conditions are also available.
Vocoder Modules
The following C and optimized assembly embedded vocoder libraries are available, as source or object modules:
Narrowband
Wideband
Applications
- WIFI phones VoWLAN
- Wireless GPRS EDGE systems
- Personal communications
- Wideband IP telephony
- Audio and video conferencing
Features
- Full and half duplex modes of operation
- Wideband audio for greater clarity
- RTP framing available for all VoIP and HD VoIP configurations
- Pass ITU test vectors
- Compliant with ITU specifications
- Optimized for high performance on leading edge DSP architectures
- 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 -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
- 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 G.168 Echo Canceller and Tone Detection/Regeneration modules
- 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.)
- Various startup procedures available (V.8 and V.8bis)
- Multiple ports can be executed on a single DSP
