Data processing: speech signal processing – linguistics – language – Audio signal time compression or expansion
Patent
1998-07-01
2000-09-05
Zele, Krista
Data processing: speech signal processing, linguistics, language
Audio signal time compression or expansion
704504, G10K 2104
Patent
active
061156884
DESCRIPTION:
BRIEF SUMMARY
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a method of and an apparatus for coding an audio signal or a plurality of audio signals, in which coded signals, produced on the basis of the audio signal or audio signals, respectively, by coding, with low quality and low bit rate, and possibly with high quality and high bit rate in addition, are transmitted to a decoder for being decoded by the same optionally with low quality or with high quality.
2. Description of Related Art
Although scalable audio coding systems in the sense of the future standard MPEG-4 are not yet available nowadays, the scalability is an essential requirement for supporting the novel functionalities of the future ISO MPEG-4 standard.
In general, the scalability is understood to be the possibility of decoding a partial set of the bit stream representing the coded audio signal so as to form a usable signal. This property is desired in particular when, for example, a data transmission channel does not make available the required entire bandwidth for transmitting a complete bit stream. Another example is incomplete decoding by a decoder of low complexity. Although continuous, complete scalability would be desirable, various discrete scalability layers are defined in practical application.
Different scalability types thus are a constituent part of the list of requirements for the future novel MPEG-4 audio standard.
First suggestions concerning a bit rate scalable system have been described in Brandenburg, H. and Grill, B., 1994, "First Ideas on Scalable Audio Coding", 9th AES Convention, San Francisco 1995, preprint No. 3924.
In the method elucidated in this technical publication, an audio signal is coded first with full bandwidth by means of an innermost so-called "layer" which is constituted by an audio codec operated with a low sampling rate. A difference signal formed by subtracting the decoded signal of the innermost layer from the initial signal or original signal is then coded in an additional audio coder or by means of two cascaded audio codecs operating on the same principle. The particular coded signals are transmitted to the decoder in a common bit stream.
A major problem of this suggested, but so far not implemented technology can be seen in that, for coding a signal of under 10 kbps, a sampling rate of 8 kHz is necessary in the first stage for obtaining reasonable results. Consequently, the entire system delay may be in the order of magnitude of one second or even more.
SUMMARY OF THE INVENTION
On the basis of this prior art, it is the object of the present invention to develop a coding method of the type set forth at the beginning in such a manner that a reduction of the delay caused by coding is obtained.
According to a first aspect, the invention provides a method of coding at least one audio signal, the method having the steps of: generating a first coded signal by coding the audio signal with a low bit rate and low delay in comparison with the delay occurring in coding the audio signal with high quality, and transmitting the first coded signal to a decoder prior to transmitting at least one additional coded signal to the decoder, which alone or together with the first coded signal provides a decoded signal with the high quality upon decoding.
According to a second aspect, the invention provides a method of coding at least one audio signal, in which coded signals, produced on the basis of the at least one audio signal by coding, with low quality and bit rate, and optionally with high quality and bit rate in addition, are transmitted to a decoder for being decoded by the same, the method having the following method steps: generating a first coded signal by coding the audio signal with a low bit rate and low delay in comparison with the delay occurring in coding the audio signal with high quality, generating a second coded signal by coding the audio signal or at least one additional signal derived from the audio signal, with a high bit rate, with the second coded signal alone or to
REFERENCES:
Brandenburg et al: "Mikroelektronik in der Audiocodierung" in:me, 1995, Nr. 1, S. 24-27.
Brandenburg, Grill: "First Ideas on Scalable Audio Coding" 97th AES-Conveon, San Francisco 1995, Vorabdruck 3924, 1994.
Brandenburg, Grill: "A Two- or Three-Stage Bit Rate Scalable Audio Coding System", Presented at the 99.sup.th Convention of AES, New York, Oct. 6-9, 1995.
Shen, A. et al: "A Robust Variable-Rate Speech Coder", 1995 International Conference on Acoustics, Speech, and Signal Processing, Detroit, MI., May 9-12, 1995.
Campbell et al: "Meeting End-to-End QoS Challenges for Scalable Flows in Heterogeneous Multimedia Environments", Technical Report, Center for Telecommunications Research, Nov. 9, 1995.
Kudumakis, Sandler: "Wavelet Packet Based Scalable Ausio Coding", 1996 IEEE International Symposium on Circuits and Systems, Atlanta, Georgia, May 12-15, 1996.
Brandenburg Karlheinz
Grill Bernhard
Seitzer Dieter
Fraunhofer-Gesellschaft zur Forderung der ange-wandten Forschung
Opsasnick Michael N.
Zele Krista
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