Ozeki VoIP SDK - Product Guide
G.722.1 is an ITU-T standard wideband audio codec that is ensure high quality for various applications like VoIP, video conferencing, teleconferencing, and Internet streaming applications. Its official name is Low-complexity coding at 24 and 32 kbps for hands-free operation in systems with low frame loss.
G.722.1 is a transform-based compressor that is optimized for speech, music and other types of audio since it is able to provide high quality experience. It uses MLT (Modulated Lapped Transform) and operates on 20 ms frames (320 samples) of audio. MLT is a critically sampled and perfect reconstruction linear transform.
Since G.722.1 is a digital wideband coder algorithm it has an audio bandwidth of 50 kHz to 7kHz and 16, 24, 32 kbps bit rates. Therefore it becomes very useful for hands-free operations in HD VoIP systems with low frame loss.
Practically, ensuring wideband coding means that the frequencies needed for speech are fully represented and this fact provides very high quality operation.
- Encoded bandwidth: ~ 50-7000 Hz
- Standardized: ITU-T 1999
- Coding type: Transform coding
- Bit rate: 24/32 kbps
- Delay (ms):
- Frame size: 20
- Lookahead: 20
- Quality: Good music performance. Scope of standard limited to hands-free and low packet loss rates. Transform coding yields poor speech quality in some operating conditions.
- MIPS: < 15
- RAM (words): 2 K
- Full and half duplex modes of operation
- Compliant with G.722.1 specification
- Optimized for high performance on leading edge DSP architectures
- Multichannel implementation
- Multi-tasking environment compatible
- Widebrand IP telephony
- Streaming audio over the Internet
- Video conferencing
- Audio conferencing
- Audio enabling your web site
Summary for G722 codec
|Algorithm||Sample Rate||Bit rate||Bits per sample||Latency||CBR||VBR||Stereo||Multi - |
|Lossy||16 kHz||24, 32 kbit/s||16 bit||40 ms||Yes||No||No||No|