A speechcoding which codes only a single speaker is currently being researched. The objective is to produce a coder which can code at a much lower bitrate than is currently available. Possible methods, including pre...
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A speechcoding which codes only a single speaker is currently being researched. The objective is to produce a coder which can code at a much lower bitrate than is currently available. Possible methods, including predictive mechanisms to reduce the bitrate, have been put forward and form the basis of future research in this area.
This paper describes a speechcoding technique that has been developed in order to provide a method of digitising speech at bitrates in the range 4. 8 to 8 kb/s, that is insensitive to the effects of acoustic backgro...
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This paper describes a speechcoding technique that has been developed in order to provide a method of digitising speech at bitrates in the range 4. 8 to 8 kb/s, that is insensitive to the effects of acoustic background noise and bit errors on the digital link. The main aim has been to develop a coding scheme which provides speech quality and robustness against noise and errors that is similar to a 16000 b/s continuously variable slope delta (CVSD) coder, but which operates at half its data rate or less. A desirable aim was to keep the complexity of the coding scheme within the scope of what could reasonably be handled by current signal processing chips or by a single custom integrated circuit. applications areas include mobile radio and small Satcomms terminals.
Source coding and encryption are linked theoretically by the aim of removing redundancy. So far, no attempt has been made to combine source coding and encryption together, except in lossless compression models. The ef...
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Source coding and encryption are linked theoretically by the aim of removing redundancy. So far, no attempt has been made to combine source coding and encryption together, except in lossless compression models. The effect of combined encoding and encryption on Analysis-by-Synthesis (AbS) LPC based techniques, a class of time domain speech compression algorithms which are widely used nowadays in commercial as well as military communication applications is considered. A novel Pre-Processing speech Scrambling Algorithm (PSSA) is proposed, which given speech, produces a scrambled signal with speech like characteristics. The resulting signal can then be compressed by a lowbitratespeech codec.
There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbitrate, typically 16 kbit/sec or below. To achieve an acceptable quality, the origi...
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There are now a number of applications, most notably streamed Internet audio, which require audio and speech to be encoded at a lowbitrate, typically 16 kbit/sec or below. To achieve an acceptable quality, the original signal is normally low-pass filtered to somewhere between 4 and 5.5 kHz before encoding. Rather than discard the upper frequency band completely, we propose encoding just the noise component of it using about 500 bits/sec. This greatly enhances contemporary music and close-microphone speech, but has little effect on classical music. The process can be used to enhance any audio or speech codec, knowing only its encoding/decoding delay.
The growing importance of research into audio to the fast-moving field of Mobile Multimedia is highlighted. Some early results from two projects at King's College which examine high quality, lowbitratecoding of...
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The growing importance of research into audio to the fast-moving field of Mobile Multimedia is highlighted. Some early results from two projects at King's College which examine high quality, lowbitratecoding of audio, both music and speech, are presented. One approach uses Wavelets, which offer advantages over polyphase filter banks in MPEG at lowbitrates. The other approach uses the long established Linear Predictive coding technique, but in a modified guise, with orders significantly higher than 10.
The authors describe a method for speechcoding based on the prototype waveform interpolation (PWI) technique proposed by Kleijn (1991). A new, variable frame length form of this technique is proposed. The basic idea ...
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The authors describe a method for speechcoding based on the prototype waveform interpolation (PWI) technique proposed by Kleijn (1991). A new, variable frame length form of this technique is proposed. The basic idea is consistent with that of the original PWI algorithm, but reduces its computational complexity. PWI methods have the potential for producing high quality speech at lowbitrates.< >
This paper briefly describes the TETRA standard, details the requirements for speech transmission in this system, describes the procedure adopted for selection of a codec algorithm and gives details of one of the code...
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This paper briefly describes the TETRA standard, details the requirements for speech transmission in this system, describes the procedure adopted for selection of a codec algorithm and gives details of one of the codec candidates.< >
The use of channel coding in lowbitratespeech coders face a performance upper bound with the limited redundancy permitted. Thus, the speech quality can degrade in bad channel transmission conditions. This paper des...
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The use of channel coding in lowbitratespeech coders face a performance upper bound with the limited redundancy permitted. Thus, the speech quality can degrade in bad channel transmission conditions. This paper describes a error control configuration which exploits source redundancy to enhance the performance of the channel decoder, possibly beyond maximum likelihood performance.< >
The author provides an overview of the standardisation activities in the field of speechcoding technology. The paper is focused primarily on narrowband applications, i.e. 8 kHz sampled signals, and its main aim is to...
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The author provides an overview of the standardisation activities in the field of speechcoding technology. The paper is focused primarily on narrowband applications, i.e. 8 kHz sampled signals, and its main aim is to give a global picture on current standardisation work in progress. As this covers a vast scope, the paper only attempts to describe those processes which will have a significant impact on the evolving personal communications expansion of the future.< >
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