摘要:
A multi-channel audio compression technology is presented that extends the range of sampling frequencies compared to existing technologies and/or lowers the noise floor while remaining compatible with those earlier generation technologies. The high-sampling frequency multi-channel audio (12) is decomposed into core audio up to the existing sampling frequencies and a difference signal up to the sampling frequencies of the next generation technologies. The core audio is encoded (18) using the first generation technology such as DTS, DOLBY AC-3 or MPEG I or MPEG II such that the encoded core bit stream (20) is fully compatible with a comparable decoder in the market. The difference signal (34) is encoded (36) using technologies that extend the sampling frequency and/or improve the quality of the core audio. The compressed difference signal (38) is attached as an extension to the core bit stream (20). The extension data will be ignored by the first generation decoders but can be decoded by the second generation decoders. By summing the decoded core and extension audio signals together (28), a second generation decoder can effectively extend the audio signal bandwidth and/or improve the signal to noise ratio beyond that available through the core decoder alone.
摘要:
A transmitting apparatus (1) has a band adaptive type input data converting section (12) and a compression encoding section (13). The band adaptive type input data converting section (12) includes a frequency band setting section for setting a frequency band for a to-be-transmitted digital sound signal, and determining a cut-off frequency and a re-sampling frequency for the digital sound signal on the basis of the set frequency band, an adaptive anti-aliasing filter for adaptively filtering the to-be-transmitted digital sound signal on the basis of the cut-off frequency, and a re-sampling section for re-sampling the digital sound signal having passed through the adaptive anti-aliasing filter, using the re-sampling frequency. The compression encoding section (13) performs compression encoding on the re-sampled digital sound signal in units of a predetermined number of data samples, thereby creating encoded data.
摘要:
Two-related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.