DIGITAL SPEECH offers Time Scale modification algorithm  for VoIP systems. This algorithm is intended for speech and audio signals acceleration/deceleration without changing of main speech parameters. Speech pitch, spectrum color aren't changed for such modification. This algorithm is intended for speech and audio acceleration/deceleration in Adaptive jitter buffer for VoIP systems. This algorithm can be used in other applications too (for example in answering machine for speech compression )

• support 8kHz and 16kHz sampling rate
• support arbitrary frame length
• compatible with any audio and speech codecs
• doesn't change speech pitch and spectrum envelope
• doesn't decrease speech and audio quality