CA2293165A1 - Method for transmitting data in wireless speech channels - Google Patents
Method for transmitting data in wireless speech channels Download PDFInfo
- Publication number
- CA2293165A1 CA2293165A1 CA002293165A CA2293165A CA2293165A1 CA 2293165 A1 CA2293165 A1 CA 2293165A1 CA 002293165 A CA002293165 A CA 002293165A CA 2293165 A CA2293165 A CA 2293165A CA 2293165 A1 CA2293165 A1 CA 2293165A1
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- speech
- vocoder
- output
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Classifications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/018—Audio watermarking, i.e. embedding inaudible data in the audio signal
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/16—Vocoder architecture
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
- G10L25/78—Detection of presence or absence of voice signals
Abstract
Non-speech information is sent in the bits allocated to one or both of a vocoder's codebooks' output by setting the gain for the corresponding codebook to zero.
By setting the gain to zero, the codebook output will not be interpreted by the receiving vocoder. In this way, it is possible to transmit additional information in a way that is totally transparent to the vocoder. Applications for this technique of sending "secret" messages include, but is not limited to, transmitting parameters for generating non-speech signals. As an example, information to generate call waiting tones, DTMF, or TTY/TDD characters can be clandestinely embedded in the compressed bit stream so that these non-speech tones can be regenerated.
By setting the gain to zero, the codebook output will not be interpreted by the receiving vocoder. In this way, it is possible to transmit additional information in a way that is totally transparent to the vocoder. Applications for this technique of sending "secret" messages include, but is not limited to, transmitting parameters for generating non-speech signals. As an example, information to generate call waiting tones, DTMF, or TTY/TDD characters can be clandestinely embedded in the compressed bit stream so that these non-speech tones can be regenerated.
Description
S. A. Benno 1 METHOD FOR TRANSMITTING DATA IN
WIRELESS SPEECH CHANNELS
Background of the Invention 1. Field of the Invention The present invention relates to telecommunications; more particularly, to transmitting data in wireless speech channels.
WIRELESS SPEECH CHANNELS
Background of the Invention 1. Field of the Invention The present invention relates to telecommunications; more particularly, to transmitting data in wireless speech channels.
2. Description of the Prior Art A voice encoder/de':oder (vocoder) is used to compress voice signals so as to reduce the transmission bandwidth over a communications channel. By reducing the bandwidth per call, it becomes possible to place more calls over the same channel. There exists a class of vocoders known as code excited lineax prediction (CELP) vocoders. In these vocoders, the speech is modeled by a series of filters. The parameters to these filters can be transmitted with much fewer bits than the original speech. It is also necessary to transmit the input (or excitation) to these filters in order to reconstruct the original speech.
Because it would require too much bandwidth to transmit the excitation directly, a crude approximation is made by replacing the excitation by a. few non-zero pulses. The locations of these pulses can be transmitted using very few trits and this crude approximation to the original excitation is adequate to reproduce high quality speech. The excitation is represented by a fixed codebook contribution and an associated gain. Also the quasi-periodicity found in speech is represented by an adaptive codebook output and an associated gain. The fixed codebook output and its associated gain, the adaptive: codebook output and its associated gain, and filter parameters (also known as linear predictive coder parameters) are transmitted to represent the encoded speech signal.
The vocoders were initially designed to compress speech by modeling its characteristics and transmitting the parameters of that model in much fewer bits than transmitting the speech itself. As wireless phones become more commonplace, people are increasingly expecting to use them for the same range of non-speech applications as they have used traditional landline phones, such as accessing voice mail and receiving call waiting tones.
Recently, the FCC has S. A. Benno 1 mandated that text-telephones for the hearing impaired (TTY/TDD) work with digital cellular phones. The problem with non-speech applications is that they do not fit the vocoder's speech model. When non-speech signals are passed through the vocoder, the decoded result it not always acceptable. The problem is further exacerbated by the fact that wireless phones operate in an error prone environment. In order to recover from transmission errors, the vocoder depends on a speech model to recover from random errors. Once again, non-speech signals do not match this model and so the reconstruction is inadequate.
Summary of the Invention The present invention sends information in the bits allocated to one or both of the codebooks' output by setting; the gain for the corresponding codebook to zero.
By setting the gain to zero, the codebook output will not be interpreted by the receiving vocoder. In this way, it is possible to transmit additional information in a way that is totally transparent to the vocoder. Applications for this technique of sending "secret" messages include, but is not limited to, transmitting parameters for generating non-speech signals. As an example, information to generate call waiting tones, DTMF, or TTY/T'DD characters can be clandestinely embedded in the compressed bit stream so that these non-speech tones can be regenerated.
Brief Description of the Drawings FIG. 1 is a block diagram of a typical vocoder;
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10; and FIG. 3 is a functional block diagram of decoder 20 of vocoder 10.
Detailed Description FIG. 1 illustrates a block diagram of a typical vocoder. Vocoder 10 receives digitized speech on input 12. The digitized speech is an analog speech signal that has been passed through an analog to digitized converter, and has been broken into frames where each frame is typically on the order of 20 nnilliseconds. The signal at input 12 is passed to encoder section 14 which encodes the speech so as decrease the amount of bandwidth used to transmit the speech.
The encoded speech is made available at output 16. 'The encoded speech is received by the decode section of a similar vocoder at the other end of a communication channel. The decoder S. A. Benno 1 at the other end of the communication channel is similar or identical to the decoder portion of vocoder 10. Encoded speech is received by vocoder 10 through input 18, and is passed to decoder section 20. Decoder section 20 uses the encoded signals received from the transmitting vocoder to produce digitized speech at output 22.
Vocoders are well known in the communications arts. For example, vocoders are described in "Speech and audio coding for wireless and network applications,"
edited by Bishnu S. Atal, Vladimir Cuperman, and Allen Gersho, 1993, by Kluwer Academic Publishers.
Vocoders are widely available and manufactured by companies such as Qualcomm Incorporated of San Diego, California, and Lucent Technologies Inc., of Murray Hill, New Jersey.
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10. A
digitized speech signal is received at input 12, and is passed to linear predictive coder 40.
Linear predictive coder 40 performs a linear predictive analysis of the incoming speech once per frame. Linear predictive analysis is well l~:nown in the art and produces a linear predictive synthesis model of the vocal tract based on the input speech signal. The linear predictive parameters or coefficients describing this model are transmitted as part of the encoded speech signal through output 16. Coder 40 uses this model to produce a residual speech signal which represents the excitation that the model uses to reproduce the input speech signal. The residual speech signal is made available at output 42. The residual speech from output 42 is provided to input 48 of open-loop pitch search unit 50 to an input of adaptive codebook unit 72 and to fixed codebook unit 82.
Impulse response unit 60 receives the linear predictive parameters from coder 40 and generates the impulse response of the model generated in coder 40. This impulse response is used in the adaptive and fixed codebook units.
Open loop pitch search unit 50 uses the residual speech signal from coder 40 to model its pitch and provides a pitch, or what is commonly called the pitch period or pitch delay signal, at output 52. The pitch delay signal from output 52 and the impulse response signal from output 64 of impulse response unit 60 are received by input 70 of adaptive codebook unit 72.
Adaptive codebook unit 72 produces a pitch gain output and a pitch index output which become part of encoded speech output 16 of vocoder 10. Output 74 of adaptive codebook 72 also provides the pitch gain and :pitch index signals to input 80 of fixed codebook unit 82.
S. A. Benno 1 Additionally, adaptive codebook 72 provides an excitation signal and an adaptive codebook target signal to input 80.
The adaptive codebook 72 produces its outputs using the digitized speech signal from input 12 and the residual speech signal produced by linear predictive coder 40. Adaptive :i codebook 72 uses the digitized speech signal and linear predictive coder 40's residual speech signal to form an adaptive c;odebook target signal. The adaptive codebook target signal is used as an input to fixed codebook 82, and as an input to a computation that produces the pitch gain, pitch index and excitation outputs of adaptive codebook unit 72. Additionally, the adaptive codebook target signal, the pitch delay signal from open loop pitch search unit 50, and the 1 (1 impulse response from impulse response unit 60 are used to produced the pitch index, the pitch gain and excitation signals which are passed to fixed codebook unit 82. The manner in which these signals are computed is well known in the vocoder art.
Fixed codebook 82 uses the inputs received from input 80 to produce a fixed gain output and a fixed index output which are used as part of the encoded speech at output 16. The 15 fixed codebook unit attempts to model the stochastic part of the linear predictive coder 40's residual speech signal. A target for the fixed codebook search is produced by determining the error between the current adaptive codebook target signal and the residual speech signal. The fixed codebook search produces the fixed gain and fixed index signal for excitation pulses so as to minimize this error. The manner in which the fixed gain and fixed index signals are 20 computed using the outputs from adaptive codebook unit 72 are well known in the vocoder art.
Switches 90 and 92 are used to send data in place of the bits that are used to send the fixed codebook output and the adaptive codebook output, respectively. When the contacts of the switches are in position "A'", the associated codebook output is replaced by data or other information and the associal:ed codebook gain is set to zero or substantially zero. As a result, 25 the scaled codebook output ~or excitation produced at a receiver will be zero or substantially zero and therefore will not have an adverse affect on the filter being used by the receiving vocoder to model the speech that is normally transmitted.
FIG. 3 illustrates a functional block diagram of decoder 20 of vocoder 10.
Encoded speech signals are received at input 18 of encoder 20. The encoded speech signals are received 30 by decoder 100. Decoder 100 produces fixed and adaptive code vectors corresponding to the fixed index and pitch index aignals, respectively. These code vectors are passed to the S. A. Benno 1 excitation construction portion of unit 110 along with the pitch gain and the fixed gain signals.
The pitch gain signal is used to scale the adaptive vector which was produced using the pitch index signal, and the fixed ;gain signal is used to scale the fixed vector which was obtained using the fixed index signali. Decoder 100 passes the linear predictive code parameters to the _'c filter or model synthesis section of unit 110. Unit 110 then uses the scaled vectors to excite the f lter that is synthesized using the linear predictive coefficients produced by linear predictive coder 40, and produces an output signal which is representative of the digitized speech originally received at input 12. Optionally, post filter 120 may be used to shape the spectrum of the digitized speech signal that is produced at output 20.
1 Ci When data rather than speech information is being transmitted, the pitch index (adaptive codebook output) and/or the fixed index (the fixed codebook output) are used to receive the data. The affect of non-data signals on the filter synthesize by unit 110 are eliminated because the gain value associated with the pitch or code index is zero.
The functional block diagrams can be implemented in various forms. Each block can be 15 implemented individually using microprocessors or microcomputers, or they can be implemented using a single microprocessor or microcomputer. It is also possible to implement each or all of the functional blocks using programmable digital signal processing devices or specialized devices received from the aforementioned manufacturers or other semiconductor manufacturers.
Because it would require too much bandwidth to transmit the excitation directly, a crude approximation is made by replacing the excitation by a. few non-zero pulses. The locations of these pulses can be transmitted using very few trits and this crude approximation to the original excitation is adequate to reproduce high quality speech. The excitation is represented by a fixed codebook contribution and an associated gain. Also the quasi-periodicity found in speech is represented by an adaptive codebook output and an associated gain. The fixed codebook output and its associated gain, the adaptive: codebook output and its associated gain, and filter parameters (also known as linear predictive coder parameters) are transmitted to represent the encoded speech signal.
The vocoders were initially designed to compress speech by modeling its characteristics and transmitting the parameters of that model in much fewer bits than transmitting the speech itself. As wireless phones become more commonplace, people are increasingly expecting to use them for the same range of non-speech applications as they have used traditional landline phones, such as accessing voice mail and receiving call waiting tones.
Recently, the FCC has S. A. Benno 1 mandated that text-telephones for the hearing impaired (TTY/TDD) work with digital cellular phones. The problem with non-speech applications is that they do not fit the vocoder's speech model. When non-speech signals are passed through the vocoder, the decoded result it not always acceptable. The problem is further exacerbated by the fact that wireless phones operate in an error prone environment. In order to recover from transmission errors, the vocoder depends on a speech model to recover from random errors. Once again, non-speech signals do not match this model and so the reconstruction is inadequate.
Summary of the Invention The present invention sends information in the bits allocated to one or both of the codebooks' output by setting; the gain for the corresponding codebook to zero.
By setting the gain to zero, the codebook output will not be interpreted by the receiving vocoder. In this way, it is possible to transmit additional information in a way that is totally transparent to the vocoder. Applications for this technique of sending "secret" messages include, but is not limited to, transmitting parameters for generating non-speech signals. As an example, information to generate call waiting tones, DTMF, or TTY/T'DD characters can be clandestinely embedded in the compressed bit stream so that these non-speech tones can be regenerated.
Brief Description of the Drawings FIG. 1 is a block diagram of a typical vocoder;
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10; and FIG. 3 is a functional block diagram of decoder 20 of vocoder 10.
Detailed Description FIG. 1 illustrates a block diagram of a typical vocoder. Vocoder 10 receives digitized speech on input 12. The digitized speech is an analog speech signal that has been passed through an analog to digitized converter, and has been broken into frames where each frame is typically on the order of 20 nnilliseconds. The signal at input 12 is passed to encoder section 14 which encodes the speech so as decrease the amount of bandwidth used to transmit the speech.
The encoded speech is made available at output 16. 'The encoded speech is received by the decode section of a similar vocoder at the other end of a communication channel. The decoder S. A. Benno 1 at the other end of the communication channel is similar or identical to the decoder portion of vocoder 10. Encoded speech is received by vocoder 10 through input 18, and is passed to decoder section 20. Decoder section 20 uses the encoded signals received from the transmitting vocoder to produce digitized speech at output 22.
Vocoders are well known in the communications arts. For example, vocoders are described in "Speech and audio coding for wireless and network applications,"
edited by Bishnu S. Atal, Vladimir Cuperman, and Allen Gersho, 1993, by Kluwer Academic Publishers.
Vocoders are widely available and manufactured by companies such as Qualcomm Incorporated of San Diego, California, and Lucent Technologies Inc., of Murray Hill, New Jersey.
FIG. 2 illustrates the major functions of encoder 14 of vocoder 10. A
digitized speech signal is received at input 12, and is passed to linear predictive coder 40.
Linear predictive coder 40 performs a linear predictive analysis of the incoming speech once per frame. Linear predictive analysis is well l~:nown in the art and produces a linear predictive synthesis model of the vocal tract based on the input speech signal. The linear predictive parameters or coefficients describing this model are transmitted as part of the encoded speech signal through output 16. Coder 40 uses this model to produce a residual speech signal which represents the excitation that the model uses to reproduce the input speech signal. The residual speech signal is made available at output 42. The residual speech from output 42 is provided to input 48 of open-loop pitch search unit 50 to an input of adaptive codebook unit 72 and to fixed codebook unit 82.
Impulse response unit 60 receives the linear predictive parameters from coder 40 and generates the impulse response of the model generated in coder 40. This impulse response is used in the adaptive and fixed codebook units.
Open loop pitch search unit 50 uses the residual speech signal from coder 40 to model its pitch and provides a pitch, or what is commonly called the pitch period or pitch delay signal, at output 52. The pitch delay signal from output 52 and the impulse response signal from output 64 of impulse response unit 60 are received by input 70 of adaptive codebook unit 72.
Adaptive codebook unit 72 produces a pitch gain output and a pitch index output which become part of encoded speech output 16 of vocoder 10. Output 74 of adaptive codebook 72 also provides the pitch gain and :pitch index signals to input 80 of fixed codebook unit 82.
S. A. Benno 1 Additionally, adaptive codebook 72 provides an excitation signal and an adaptive codebook target signal to input 80.
The adaptive codebook 72 produces its outputs using the digitized speech signal from input 12 and the residual speech signal produced by linear predictive coder 40. Adaptive :i codebook 72 uses the digitized speech signal and linear predictive coder 40's residual speech signal to form an adaptive c;odebook target signal. The adaptive codebook target signal is used as an input to fixed codebook 82, and as an input to a computation that produces the pitch gain, pitch index and excitation outputs of adaptive codebook unit 72. Additionally, the adaptive codebook target signal, the pitch delay signal from open loop pitch search unit 50, and the 1 (1 impulse response from impulse response unit 60 are used to produced the pitch index, the pitch gain and excitation signals which are passed to fixed codebook unit 82. The manner in which these signals are computed is well known in the vocoder art.
Fixed codebook 82 uses the inputs received from input 80 to produce a fixed gain output and a fixed index output which are used as part of the encoded speech at output 16. The 15 fixed codebook unit attempts to model the stochastic part of the linear predictive coder 40's residual speech signal. A target for the fixed codebook search is produced by determining the error between the current adaptive codebook target signal and the residual speech signal. The fixed codebook search produces the fixed gain and fixed index signal for excitation pulses so as to minimize this error. The manner in which the fixed gain and fixed index signals are 20 computed using the outputs from adaptive codebook unit 72 are well known in the vocoder art.
Switches 90 and 92 are used to send data in place of the bits that are used to send the fixed codebook output and the adaptive codebook output, respectively. When the contacts of the switches are in position "A'", the associated codebook output is replaced by data or other information and the associal:ed codebook gain is set to zero or substantially zero. As a result, 25 the scaled codebook output ~or excitation produced at a receiver will be zero or substantially zero and therefore will not have an adverse affect on the filter being used by the receiving vocoder to model the speech that is normally transmitted.
FIG. 3 illustrates a functional block diagram of decoder 20 of vocoder 10.
Encoded speech signals are received at input 18 of encoder 20. The encoded speech signals are received 30 by decoder 100. Decoder 100 produces fixed and adaptive code vectors corresponding to the fixed index and pitch index aignals, respectively. These code vectors are passed to the S. A. Benno 1 excitation construction portion of unit 110 along with the pitch gain and the fixed gain signals.
The pitch gain signal is used to scale the adaptive vector which was produced using the pitch index signal, and the fixed ;gain signal is used to scale the fixed vector which was obtained using the fixed index signali. Decoder 100 passes the linear predictive code parameters to the _'c filter or model synthesis section of unit 110. Unit 110 then uses the scaled vectors to excite the f lter that is synthesized using the linear predictive coefficients produced by linear predictive coder 40, and produces an output signal which is representative of the digitized speech originally received at input 12. Optionally, post filter 120 may be used to shape the spectrum of the digitized speech signal that is produced at output 20.
1 Ci When data rather than speech information is being transmitted, the pitch index (adaptive codebook output) and/or the fixed index (the fixed codebook output) are used to receive the data. The affect of non-data signals on the filter synthesize by unit 110 are eliminated because the gain value associated with the pitch or code index is zero.
The functional block diagrams can be implemented in various forms. Each block can be 15 implemented individually using microprocessors or microcomputers, or they can be implemented using a single microprocessor or microcomputer. It is also possible to implement each or all of the functional blocks using programmable digital signal processing devices or specialized devices received from the aforementioned manufacturers or other semiconductor manufacturers.
Claims (5)
1. A method for transmitting non-speech information over a speech channel, CHARACTERIZED BY the steps of:
transmitting non-speech information in place of pitch index information; and transmitting a pitch gain value having a value of substantially zero.
transmitting non-speech information in place of pitch index information; and transmitting a pitch gain value having a value of substantially zero.
2. The method of claim 1, CHARACTERIZED IN THAT the non-speech information is DTMF information.
3. The method of claim 1, CHARACTERIZED IN THAT the non-speech information is TTY/TDD information.
4. A method for transmitting non-speech information over a speech channel, CHARACTERIZED BY the steps of:
transmitting first non-speech information in place of fixed index information;
and transmitting a index gain value having a value of substantially zero.
transmitting first non-speech information in place of fixed index information;
and transmitting a index gain value having a value of substantially zero.
5. The method of claim 4, further CHARACTERIZED BY the steps of:
transmitting second non-speech information in place of pitch index information; and transmitting a pitch gain value having a value of substantially zero.
transmitting second non-speech information in place of pitch index information; and transmitting a pitch gain value having a value of substantially zero.
Applications Claiming Priority (2)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| US22810299A | 1999-01-11 | 1999-01-11 | |
| US09/228,102 | 1999-01-11 |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CA2293165A1 true CA2293165A1 (en) | 2000-07-11 |
Family
ID=22855803
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CA002293165A Abandoned CA2293165A1 (en) | 1999-01-11 | 1999-12-30 | Method for transmitting data in wireless speech channels |
Country Status (7)
| Country | Link |
|---|---|
| EP (1) | EP1020848A2 (en) |
| JP (1) | JP2000209663A (en) |
| KR (1) | KR20000053407A (en) |
| CN (1) | CN1262577A (en) |
| AU (1) | AU6533799A (en) |
| BR (1) | BR0000002A (en) |
| CA (1) | CA2293165A1 (en) |
Families Citing this family (13)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| AU2002228646A1 (en) * | 2000-11-07 | 2002-05-21 | Ericsson Inc. | Method of and apparatus for detecting tty type calls in cellular systems |
| US7310596B2 (en) | 2002-02-04 | 2007-12-18 | Fujitsu Limited | Method and system for embedding and extracting data from encoded voice code |
| JP4330346B2 (en) * | 2002-02-04 | 2009-09-16 | 富士通株式会社 | Data embedding / extraction method and apparatus and system for speech code |
| US7932851B1 (en) * | 2002-10-15 | 2011-04-26 | Itt Manufacturing Enterprises, Inc. | Ranging signal structure with hidden acquisition code |
| US7970606B2 (en) | 2002-11-13 | 2011-06-28 | Digital Voice Systems, Inc. | Interoperable vocoder |
| US7634399B2 (en) | 2003-01-30 | 2009-12-15 | Digital Voice Systems, Inc. | Voice transcoder |
| EP1455509A3 (en) * | 2003-03-03 | 2005-01-05 | FREQUENTIS GmbH | Method and system for speech recording |
| US8359197B2 (en) | 2003-04-01 | 2013-01-22 | Digital Voice Systems, Inc. | Half-rate vocoder |
| FR2859566B1 (en) * | 2003-09-05 | 2010-11-05 | Eads Telecom | METHOD FOR TRANSMITTING AN INFORMATION FLOW BY INSERTION WITHIN A FLOW OF SPEECH DATA, AND PARAMETRIC CODEC FOR ITS IMPLEMENTATION |
| US7752039B2 (en) | 2004-11-03 | 2010-07-06 | Nokia Corporation | Method and device for low bit rate speech coding |
| DE102007007627A1 (en) * | 2006-09-15 | 2008-03-27 | Rwth Aachen | Method for embedding steganographic information into signal information of signal encoder, involves providing data information, particularly voice information, selecting steganographic information, and generating code word |
| US8036886B2 (en) | 2006-12-22 | 2011-10-11 | Digital Voice Systems, Inc. | Estimation of pulsed speech model parameters |
| US11270714B2 (en) | 2020-01-08 | 2022-03-08 | Digital Voice Systems, Inc. | Speech coding using time-varying interpolation |
-
1999
- 1999-12-17 AU AU65337/99A patent/AU6533799A/en not_active Abandoned
- 1999-12-30 CA CA002293165A patent/CA2293165A1/en not_active Abandoned
-
2000
- 2000-01-03 BR BR0000002-7A patent/BR0000002A/en not_active Application Discontinuation
- 2000-01-06 EP EP00300042A patent/EP1020848A2/en not_active Withdrawn
- 2000-01-07 KR KR1020000000557A patent/KR20000053407A/en not_active Application Discontinuation
- 2000-01-10 CN CN00101021A patent/CN1262577A/en active Pending
- 2000-01-11 JP JP2766A patent/JP2000209663A/en active Pending
Also Published As
| Publication number | Publication date |
|---|---|
| JP2000209663A (en) | 2000-07-28 |
| AU6533799A (en) | 2000-07-13 |
| BR0000002A (en) | 2002-01-02 |
| CN1262577A (en) | 2000-08-09 |
| KR20000053407A (en) | 2000-08-25 |
| EP1020848A2 (en) | 2000-07-19 |
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Legal Events
| Date | Code | Title | Description |
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| EEER | Examination request | ||
| FZDE | Discontinued |