VOCAL Technologies speech coder software libraries include a complete range of voice compression algorithms optimized for execution on ANSI C and leading DSP architectures (TI, ADI, AMD, ARM, MIPS, CEVA, LSI Logic ZSP, etc.). Native Libraries are available for VoIP applications using Android, iOS, and other mobile devices. VOCAL’s vocoder software may be licensed as source code or binary with custom solutions available to meet unique application requirements.
ITU speech compression software algorithms were originally designed to minimize bandwidth usage in the telephone network. Voice codecs were also developed for specific applications, e.g. video conferencing. With the advent of Voice over IP, most of these legacy vocoders have been retrofitted with new features such as silence suppression and packet loss concealment. VOCAL’s speech coder implementations are fully compliant with ITU standards and incorporate the latest features.
Many GSM speech compression algorithms were developed by ETSI for the European market, standardized by the mobile telephone industry and then deployed world-wide, albeit not the United States. These voice codecs were limited by early handheld technology and channel capacity which have significantly improved over the years. VOCAL’s implementations of these GSM vocoders are fully compliant with ETSI standards.
With the broad deployment of Voice over IP, wideband speech coders have been developed for HD audio with a bandwidth of 50 Hz to 7 kHz for a fuller listening experience. Legacy telephony equipment was limited to a range of 200 to 3400 Hz. The the natural human listening range covers 20 Hz to 20,000 Hz (20 KHz) with human speech ranging up to 15 kHz. VOCAL’s implementation of wideband voice codecs allow developers to take full advantage of HD audio.
Although Android provides basic speech compression capabilities, VOCAL provides a range of standard voice codecs for developers to meet unique user application requirements. These vocoders are optimized for performance on the Android platform, may be accessed using JNI and include the ITU, GSM, Wideband, and Other speech coders.
VOCAL also provides a range of standard voice compression software algorithms in addition to the basic iOS voice codecs for developers to create customized application. These vocoders are accessible using objective C, optimized for performance on the iOS platform, and include ITU, GSM, Wideband, and Other speech coders.
Special purpose algorithms have been developed by government organizations and private industry to meet specific objectives. Several voice compression software algorithms have been developed for specific applications and challenging environments, while other vocoders have been developed as open source for general use.
Each speech coder may or may not be appropriate for a function and/or platform. Reference MIPS/memory requirements as well as PSQM/PSQM+ values for different voice coders under various network conditions are available from VOCAL for comparison. When evaluating a particular voice codec for a specific application, please contact us for the most current information.
The following C and optimized assembly embedded vocoder and associated libraries are available from VOCAL as source or object modules:
- Voice Over Network Framework – RTP Packetization, AAL2 Framing
- ITU Speech Coders (Audio Examples)
- G.711 - 64k bps PCM (A-law or μ-law form)
- G.722 - 7 kHz audio coding within 64 kbit/s (SB-ADPCM)
- G.722.1 - 24k and 32k bps, 7 kHz audio
- G.722.2 - Adaptive Multi-Rate Wideband (GSM AMR-WB)
- G.723.1 - 5 1/3k and 6.4k bps ACELP/MP-MLQ
- G.726 - 16k, 24k, 32k and 40k bps ADPCM
- G.727 - 5, 4, 3 and 2-bits sample Embedded ADPCM
- G.728 - 16k bps LD-CELP
- G.729/G.729A - 8k bps CS-ACELP
- G.729 Annex B - Silence Detection
- GSM Speech Coders (Audio Examples)
- Wideband Speech Coders (Audio Examples)
- Other Speech Coders (Audio Examples)
- Voice Activity Detection (VAD)
- Packet Loss Concealment (PLC)
- Adaptive Jitter
- Voice/Facsimile/Data Modem Detection
- WIFI phones VoWLAN
- Wireless GPRS EDGE systems
- Personal communications
- Wideband IP telephony
- Audio and video conferencing
- Mobile communications
- Fax over IP/Fax Relay/Transcoding
- Full and half duplex modes of operation
- Pass ITU test vectors
- Compliant with ITU specifications
- Optimized for high performance on leading edge DSP architectures
- Multichannel implementation
- Multi-tasking environment compatible
- 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
- Transcoding/transcode between vocoders
- Common speech compression frame stream interface to support systems with multiple speech coders
- Dynamic speech coder 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