Telephonic communications – Audio message storage – retrieval – or synthesis – Message signal analysis
Reexamination Certificate
1998-05-13
2001-09-25
Tsang, Fan (Department: 2645)
Telephonic communications
Audio message storage, retrieval, or synthesis
Message signal analysis
C379S093080, C379S127030, C379S142030, C379S088210, C379S088200
Reexamination Certificate
active
06295340
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to voice messaging systems. More particularly, it relates to voice messaging systems wherein the speech coding used to code incoming voice messages is changeable based on incoming call related information.
2. Description of Related Art
Voice messaging has become an everyday requirement in today's society. Early voice messaging apparatus comprised magnetic cassette tapes which recorded a significant amount of voice messages, e.g., 60 minutes of voice messages. However, cassette tapes were disadvantageous because of the mechanics and time required to fast-forward and rewind the cassette tapes to the storage points of individual messages.
More recently, particularly as the size of memory has increased in density while at the same time decreased in price, digital voice messaging systems have become commonplace. Digital voice messaging systems store incoming voice messages in digital memory, reducing the mechanics and cost, and increasing the reliability over conventional analog voice messaging systems.
It is commonly known to encode voice messages for storage in digital voice messaging systems to maximize the length of voice messages which may be stored in a finite amount of memory. Various conventional voice compression techniques exist, many having various effective data rates. For instance, linear predictive coding (LPC), code-excited linear predictive (CELP) coding, RPE-LTP, or adaptive differential pulse code modulation (ADPCM) are examples of techniques, any one of which may be utilized by conventional digital voice messaging systems, and any one of which may be implemented at varying effective data rates. These coding techniques result in a compression of the data yielding a reduction in the data rate necessary to represent the underlying voice message. The effective data rate resulting from a general linear approach to digitizing toll quality speech (e.g., 8 KHz, 8-bit sampling, i.e., 64 Kb/s) with respect to an effective data rate resulting from a coding technique such as CELP, LPC, or ADPCM forms a compression ratio of the coding. These compression techniques have been developed for use with digital voice messaging systems which in general reduce the number of bits necessary to represent voice message data. Although it is recognized that the compression techniques are improved over time to better approach toll quality speech, in general the greater the compression ratio, the greater the degradation in the quality of the speech stored in voice memory. “Compression ratio” as used herein refers to the ratio of the number of bits representing the voice message in linear form divided by the number of bits used to represent the same voice message in compressed or coded form.
FIG. 6
shows a prior art voice messaging apparatus such as a digital telephone answering device (TAD)
11
. Although described with respect to embodiments relating to a TAD, the present invention is equally applicable to a voice mail type system and voice messaging systems in general.
In
FIG. 6
, a signal from a microphone or other analog signal source
106
is input to a codec, analog-to-digital converter, or other digitizing device in a voice recorder/playback module
20
for conversion to m-law or A-law pulse code modulated (PCM) data. The m-law or A-law PCM data is encoded and compressed by a variable compression ratio encoding routine in the processor
18
. For playback, a variable compression ratio decoding routing in the processor
18
together with a codec, digital-to-analog converter, or similar device in the voice recorder/playback module
20
converts voice message data retrieved from voice memory into an analog signal for output to speaker
108
.
While control of the TAD
11
and handling of voice processing is shown in
FIG. 6
as being handled in processor
18
, a separate digital signal processor may be utilized to handle the compression and other voice processing tasks. The processor
18
controls the encoding and decoding tasks performing the compression and decompression of the voice message data. Processor
18
also controls storage of the compressed (encoded) speech data into voice memory
100
, and controls the retrieval of compressed speech data from the voice memory
100
.
The TAD
11
further includes an alpha-numeric keypad
36
and display
27
as are known in the art. The display
27
displays call related information regarding an incoming call, e.g., a caller's telephone number and household name, received from a central office
13
via a telephone line interface
48
and call related information detector/receiver
12
.
However, conventional voice messaging systems are limited by the compression rate used and the amount of available message memory. Messages which are received after the memory becomes full, such as after a long vacation, are conventionally refused.
Thus, there is a need to maximize the available message memory for use beyond that which is conventionally available such that more messages may be stored.
SUMMARY OF THE INVENTION
In accordance with the principles of the present invention, a variable voice compression ratio device in a voice messaging device comprises a processor, a call related information detector/receiver, and a call related information/coding database to associate pre-set call related information with one of a plurality of voice compression ratios. The processor compares incoming call related information to the pre-set call related information to determine the desired voice compression ratio for storage of a voice message corresponding to the incoming call.
In another aspect of the invention, a variable voice compression ratio device comprises a memory containing a plurality of voice compression ratio values each associated with respective call related information. A call related information receiver receives incoming call related information, and a processor compares the received call related information with the respective call related information contained in the memory to identify one of the plurality of voice compression ratio values.
A method of setting an effective data rate for storage of a voice message for an incoming call in accordance with the present invention provides detection and reception of call related information relating to the incoming call. The detected and received call related information is compared with a memory containing pre-set call related information associated with respective effective data rates. An effective data rate is then set for storage of the voice message of the incoming call based on the comparison.
REFERENCES:
patent: 4582956 (1986-04-01), Doughty
patent: 5506872 (1996-04-01), Mohler
patent: 5524140 (1996-06-01), Klausner et al.
patent: 5881104 (1999-03-01), Akahane
patent: 5978757 (1999-11-01), Newton
patent: 6026152 (2000-02-01), Cannon et al.
patent: 6067349 (2000-05-01), Suder et al.
Cannon Joseph M.
Iyengar Vasu
Johanson James A.
Bollman William H.
Escalante Ovidio
Lucent Technologies - Inc.
Tsang Fan
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