Codebook structure for changeable pulse multimode speech coding

Details Codebook structure for changeable pulse multimode speech coding Codebook structure for changeable pulse multimode speech coding

2000-09-15

2003-04-29

To, Doris H. (Department: 2654)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

C704S219000, C704S212000, C704S227000

Reexamination Certificate

active

06556966

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Technical Field
This invention relates to speech communication systems and, more particularly, to systems and methods for digital speech coding.
2. Related Art
One prevalent mode of human communication involves the use of communication systems. Communication systems include both wireline and wireless radio systems. Wireless communication systems electrically connect with the landline systems and communicate using radio frequency (RF) with mobile communication devices. Currently, the radio frequencies available for communication in cellular systems, for example, are in the frequency range centered around 900 MHz and in the personal communication services (PCS) frequency range centered around 1900 MHz. Due to increased traffic caused by the expanding popularity of wireless communication devices, such as cellular telephones, it is desirable to reduce bandwidth of transmissions within the wireless systems.
Digital transmission in wireless radio telecommunications is increasingly being applied to both voice and data due to noise immunity, reliability, compactness of equipment and the ability to implement sophisticated signal processing functions using digital techniques. Digital transmission of speech signals involves the steps of: sampling an analog speech waveform with an analog-to-digital converter, speech compression (encoding), transmission, speech decompression (decoding), digital-to-analog conversion, and playback into an earpiece or a loudspeaker. The sampling of the analog speech waveform with the analog-to-digital converter creates a digital signal. However, the number of bits used in the digital signal to represent the analog speech waveform creates a relatively large bandwidth. For example, a speech signal that is sampled at a rate of 8000 Hz (once every 0.125 ms), where each sample is represented by 16 bits, will result in a bit rate of 128,000 (16×8000) bits per second, or 128 kbps (kilo bits per second).
Speech compression reduces the number of bits that represent the speech signal, thus reducing the bandwidth needed for transmission. However, speech compression may result in degradation of the quality of decompressed speech. In general, a higher bit rate will result in higher quality, while a lower bit rate will result in lower quality. However, speech compression techniques, such as coding techniques, can produce decompressed speech of relatively high quality at relatively low bit rates. In general, low bit rate coding techniques attempt to represent the perceptually important features of the speech signal, with or without preserving the actual speech waveform.
Typically, parts of the speech signal for which adequate perceptual representation is more difficult or more important (such as voiced speech, plosives or voice onsets) are coded and transmitted using a higher number of bits. Parts of the speech signal for which adequate perceptual representation is less difficult or less important (such as unvoiced, or the silence between words) are coded with a lower number of bits. The resulting average bit rate for the speech signal will be relatively lower than would be the case for a fixed bit rate that provides decompressed speech of similar quality.
These speech compression techniques have resulted in lowering the amount of bandwidth used to transmit a speech signal. However, further reduction in bandwidth is important in a communication system for a large number of users. Accordingly, there is a need for systems and methods of speech coding that are capable of minimizing the average bit rate needed for speech representation, while providing high quality decompressed speech.
SUMMARY
The invention provides a way to construct an efficient codebook structure and a fast search approach, which in one example are used in an SMV system. The SMV system varies the encoding and decoding rates in a communications device, such as a mobile telephone, a cellular telephone, a portable radio transceiver or other wireless or wire line communication device. The disclosed embodiments describe a system for varying the rates and associated bandwidth in accordance with an signal from an external source, such as the communication system with which the mobile device interacts. In various embodiments, the communications system selects a mode for the communications equipment using the system, and speech is processed according to that mode.
One embodiment of a speech compression system includes a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec each capable of encoding and decoding speech signals. The speech compression system performs a rate selection on a frame by frame basis of a speech signal to select one of the codecs. The speech compression system then utilizes a fixed codebook structure with a plurality of subcodebooks. A search routine selects a best codevector from among the codebooks in encoding and decoding the speech. The search routine is based on minimizing an error function in an iterative fashion.
Accordingly, the speech coder is capable of selectively activating the codecs to maximize the overall quality of a reconstructed speech signal while maintaining the desired average bit rate. Other systems, methods, features and advantages of the invention will be or will become apparent to one with skill in the art upon examination of the following figures and detailed description. It is intended that all such additional systems, methods, features and advantages included within this description be within the scope of the invention, and be protected by the accompanying claims.


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