Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-12-08
2004-10-19
McFadden, Susan (Department: 2655)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S205000, C704S208000
Reexamination Certificate
active
06807526
ABSTRACT:
FIELD OF INVENTION
The present invention relates to a method of and apparatus for processing at least one coded binary audio stream organized into frames. This or these streams are obtained by, on the one hand, frequency type coding algorithms using psychoacoustic characteristics of the human ear to reduce throughput and, on the other hand, a quantization of the thus-coded signals. The invention is particularly applicable when no bit allocation data implemented during the quantization is explicitly present in the audio streams considered.
BACKGROUND ART
One of the main problems to be resolved in processing coded audio streams is reducing the computing cost for such processing. Generally, such processing is implemented in the time domain so it is necessary to convert audio streams from the frequency domain to the time domain then, after processing the time streams, convert back from the time domain to the frequency domain. These conversions cause algorithmic times and greatly increase computing costs, which might be onerous.
In particular, in the case of teleconferencing, attempts have been made to reduce overall communication time and thus increase its quality in terms of interactivity. The problems mentioned above are even more serious in the case of teleconferencing because of the high number of accesses that a multipoint control unit might provide.
For teleconferencing, audio streams can be coded using various kinds of standardized coding algorithms. Thus, the H.320 standard, specific to transmission on narrow band ISDN, specifies several coding algorithms (G.711, G.722, G.728). Likewise, standard H.323 so specifies several coding algorithms (G.723.1, G.729 and MPEG-1).
Moreover, in high-quality teleconferencing, standard G.722 specifies a coding algorithm that operates on a 7 kHz bandwidth, subdividing the spectrum into two subbands. ADPCM type coding is then performed for the signal in each band.
To solve the problem and the complexity introduced by the banks of quadrature mirror filters, at the multipoint control unit level, Appendix I of Standard G.722 specifies a direct recombination method based on subband signals. This method consists of doing an ADPCM decoding of two samples from the subbands of each input frame of the multipoint control unit, summing all the input channels involved and finally doing an ADPCM coding before building the output frame.
One solution suggested to reduce complexity is to restrict the number of decoders at the multipoint control unit level and thus combine the coded audio streams on only a part of the streams received. There are several strategies for determining the input channels to consider. For example, combination is done on the N′ signals with the strongest gains, where N′ is predefined and fixed, and where the gain is read directly from input code words. Another example is doing the combining only on the active streams although the number of inputs considered is then variable.
It is to be noted that these approaches do not solve the time reduction problem.
SUMMARY OF THE INVENTION
The purpose of this invention is to provide a new and improved method of and apparatus for processing at least one coded binary audio stream making it possible to solve the problems mentioned above.
Such a process can be used to transpose an audio stream coded at a first throughput into another stream at a second throughput. It can also be used to combine several coded audio streams, for example, in an audio teleconferencing system.
A possible application for the process of this invention involves teleconferencing, mainly, in the case of a centralized communication architecture based on a multipoint control unit (MCU) which plays, among other things, the role of an audio bridge that combines (or mixes) audio streams then routes them to the terminals involved.
It will be noted, however, that the method and apparatus of this invention can be applied to a teleconferencing system whose architecture is of the mesh type, i.e., when terminals are point-to-point linked.
Other applications might be envisaged, particularly in other multimedia contexts. This is the case, for example, with accessing database servers containing audio objects to construct virtual scenes.
Sound assembly and editing, which involves manipulating one or more compressed binary streams to produce a new one is another area in which this invention can be applied.
Another application for this invention is transposing a stream of audio signals coded at a first throughput into another stream at a second throughput. Such an application is interesting when there is transmission through different heterogeneous networks where the throughput must be adapted to the bandwidth provided by the transmission environment used. This is the case for networks where service quality is not guaranteed (or not reliable) or where allocation of the bandwidth depends on traffic conditions. A typical example is the passage from an Intranet environment (Ethernet LAN at 10 Mbits/s, for example) where the bandwidth limitation is less severe, to a more saturated network (Internet). The new H.323 teleconferencing standard allowing interoperability among terminals on different kinds of networks (LAN for which QoS is not guaranteed, NISDN, BISDN, GSTN, . . . ) is another application area. Another interesting case is when audio servers are accessed (audio on demand, for example). Audio data are often stored in coded form but with a sufficiently low compression rate to maintain high quality, since transmission over a network might need another reduction in throughput.
The invention thus concerns a method of and apparatus for processing at least one coded binary audio stream organized as frames formed from digital audio signals which were coded by first converting them from the time domain to the frequency domain in order to calculate transform coefficients then quantizing and coding these transform coefficients based on a set of quantizers determined from a set of selection parameters that are used to select said quantizers, said selection parameters being also present in the frames.
Said method comprises: 1) a step of recovering the transform coefficients which comprises a decoding step and a dequantifying step for decoding and then dequantify the frames based on a set of quantifiers as determined from said selection parameters included in said frames of at least said coded binary audio stream, 2) a step of processing the transform coefficients thus recovered in the frequency domain and, 3) a step of supplying the processed frames to a subsequent utilization step.
According to a first implementation mode, the subsequent utilization step, called recoding step, partially recodes the frames thus processed in a step involving requantization and then recoding of the thus-processed transform coefficients.
According to another characteristic of the invention, the processing step 2) involves summing the transform coefficients produced by the recovering step 1) from the different audio streams and said recoding step involves requantizing, and then recoding the summed transform coefficients.
This described process can be performed in processing stages of a multi-terminal teleconferencing system. In such a case, The processing step 2) involves summing the transform coefficients produced by the recovering step 1) from the different audio streams, said recoding step involves, for a given terminal, subtracting the transform coefficient from said terminal to the summed transform coefficients, and requantizing and then recoding the resulting transform coefficients.
According to another implementation mode of the invention, the subsequent utilization step is a frequency domain to time domain conversion step for recovering the audio signal. Such a conversion process is performed, for example, in a multi-terminal audioconferencing system. The processing step involves summing the transform coefficients produced by the partial decoding of the frame streams coming from said terminals.
According to another characteristic of the invention
Lamblin Claude
Mahieux Yannick
Touimi Abdellatif Benjelloun
France Telecom S.A.
Lowe Hauptman & Gilman & Berner LLP
McFadden Susan
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