Source: http://www.google.com/patents/US7848922?dq=7222078
Timestamp: 2015-04-27 21:44:19
Document Index: 675330681

Matched Legal Cases: ['Application No. 60', 'Application No. 60', 'art 410', 'art 410', 'art3', 'art 3', 'art3', 'art 3']

Patent US7848922 - Method and apparatus for a thin audio codec - Google PatentsSearch Images Maps Play YouTube News Gmail Drive More »Sign inAdvanced Patent SearchPatentsAn apparatus and method for encoding and decoding a voice signal. The apparatus includes an encoder configured to generate an output bitstream signal from an input voice signal. The output bitstream signal is associated with at least a first standard of a first plurality of CELP voice compression standards....http://www.google.com/patents/US7848922?utm_source=gb-gplus-sharePatent US7848922 - Method and apparatus for a thin audio codecAdvanced Patent SearchPublication numberUS7848922 B1Publication typeGrantApplication numberUS 11/890,263Publication dateDec 7, 2010Filing dateAug 2, 2007Priority dateOct 17, 2002Fee statusPaidAlso published asUS7254533Publication number11890263, 890263, US 7848922 B1, US 7848922B1, US-B1-7848922, US7848922 B1, US7848922B1InventorsMarwan A. Jabri, Nicola Chong-White, Jianwei WangOriginal AssigneeJabri Marwan A, Nicola Chong-White, Jianwei WangExport CitationBiBTeX, EndNote, RefManPatent Citations (14), Non-Patent Citations (28), Referenced by (6), Classifications (9), Legal Events (2) External Links: USPTO, USPTO Assignment, EspacenetMethod and apparatus for a thin audio codec
US 7848922 B1Abstract
An apparatus and method for encoding and decoding a voice signal. The apparatus includes an encoder configured to generate an output bitstream signal from an input voice signal. The output bitstream signal is associated with at least a first standard of a first plurality of CELP voice compression standards. Additionally, the apparatus includes a decoder configured to generate an output voice signal from an input bitstream signal. The input bitstream signal is associated with at least a first standard of a second plurality of CELP voice compression standards. The CELP encoder includes a plurality of codec-specific encoder modules. Additionally, the CELP encoder includes a plurality of generic encoder modules. The CELP decoder includes a plurality of codec-specific decoder modules. Additionally, the CELP decoder includes a plurality of generic decoder modules.
1. An apparatus for encoding an audio signal, the apparatus comprising:
an encoder configured to generate an output bitstream signal from an input audio signal, the output bitstream signal associated with at least a first standard of a plurality of audio compression standards, the encoder comprising:
a plurality of codec-specific encoder modules, at least one of the plurality of codec-specific encoder modules including at least a first table or a first function, the first table or the first function associated with only a second standard of the plurality of audio compression standards; and
a plurality of generic encoder modules, at least one of the plurality of generic encoder modules including at least a second table or a second function, the second table or the second function associated with at least a third standard and a fourth standard of the plurality of audio compression standards, the third standard being different from the fourth standard.
2. The apparatus of claim 1 wherein the plurality of generic encoder modules comprises:
a first common functions library including at least the second function; and
a first common tables library including at least the second table.
3. The apparatus of claim 1 wherein the plurality of codec-specific encoder modules comprise:
a pre-processing module configured to process the audio signal for encoding;
an excitation generation module configured to generate an excitation signal by filtering the audio signal by the short-term prediction filter;
a fixed codebook module configured to determine fixed codebook vectors and a fixed codebook gain; and
a bitstream packing module including at least one bitstream packing routine and configured to generate the output bitstream signal based on at least one or more codec-specific parameters associated with at least the first standard of the plurality of audio compression standards.
4. The apparatus of claim 1 wherein the first standard of the plurality of audio compression standards is the same as the second standard of the plurality of audio compression standards.
5. The apparatus of claim 1 wherein the first standard of the plurality of audio compression standards is the same as the third standard or the fourth standard of the plurality of audio compression standards.
6. The apparatus of claim 1 further comprising a second encoder configured to generate a second output bitstream signal from the input audio signal, the second output bitstream signal associated with at least another standard of the plurality of audio compression standards, the another standard being different from the first standard.
7. An apparatus for decoding an audio signal, the apparatus comprising:
a decoder configured to generate an output audio signal from an input bitstream signal, the input bitstream signal associated with at least a first standard of a plurality of audio compression standards, wherein the decoder comprises:
a plurality of codec-specific decoder modules, at least one of the plurality of codec-specific decoder modules including at least a third table or a third function, the third table or the third function associated with only a second standard of the plurality of audio compression standards; and
a plurality of generic decoder modules, at least one of the plurality of generic decoder modules including at least a fourth table or a fourth function, the fourth table or the fourth function associated with at least a third standard and a fourth standard of the plurality of audio compression standards, the third standard being different from the fourth standard.
8. The apparatus of claim 7 wherein the generic decoder modules comprise:
a second common functions library including at least the fourth function; and
a second common tables library including at least the fourth table.
9. The apparatus of claim 7 wherein the plurality of codec-specific decoder modules comprise:
a bitstream unpacking module including at least one bitstream unpacking routine and configured to decode the input bitstream signal and generate codec-specific parameters;
a synthesis module configured to filter the excitation signal and generate a reconstructed audio signal; and
a post-processing module configured to improve a perceptual quality of the reconstructed audio signal.
10. The apparatus of claim 7 wherein the first standard is the same as the second standard.
11. The apparatus of claim 7 wherein the first standard is the same as the third standard or the fourth standard.
processing the input audio signal, wherein processing the input audio signal uses at least a first common functions library and at least a first common tables library, the first common functions library including a first function and the first common tables library including a first table, wherein the first function and the first table are associated with at least a second standard and a third standard of the plurality of audio compression standards, the second standard being different from the third standard; and
generating an output bitstream signal based on at least information associated with the input audio signal, the output bitstream signal associated with at least a first standard of a plurality of audio compression standards, wherein generating an output signal comprises:
generating a first plurality of codec-specific parameters based on at least information associated with the input audio signal; and
packing the first plurality of codec-specific parameters to the output bitstream signal.
13. The method of claim 12 wherein the first plurality of codec-specific parameters comprise a linear prediction parameter, an adaptive codebook lag, an adaptive codebook gain, a fixed codebook index, and a fixed codebook gain.
14. The method of claim 13 wherein the linear prediction parameter comprises a line spectral frequency.
15. The method of claim 12 wherein the generating a first plurality of codec-specific parameters comprises:
filtering the input audio signal by a short-term prediction filter;
determining an index of a fixed codebook vector associated with a fixed codebook target signal; and
16. The method of claim 12 wherein the first standard is the same as the second standard or the third standard.
17. The method of claim 12 further comprising generating a second output bitstream signal based on at least information associated with the input audio signal, the output bitstream signal associated with at least another standard of the plurality of audio compression standards, the another standard being different from the first standard.
18. A method for decoding an audio signal, the method comprising:
processing the input bitstream signal, wherein processing the input bitstream signal uses at least a second common functions library and a second common tables library, the second common functions library including a second function and the second common tables library including a second table; wherein the second function and the second table are associated with at least a second standard and a third standard of the plurality of audio compression standards, the second standard being different from the third standard; and
generating an output audio signal based on at least information associated with the input bitstream signal, the output audio signal associated with at least a first standard of a plurality of audio compression standards, wherein generating an output audio signal comprises:
unpacking the input bitstream signal; and
decoding a second plurality of codec-specific parameters to produce an output audio signal.
19. The method of claim 18 wherein decoding a second plurality of codec-specific parameters comprises:
generating an intermediate audio signal; and
processing the intermediate audio signal to improve a perceptual quality.
20. The method of claim 18 wherein the first standard is the same as the second standard or the third standard.
This application is a continuation of U.S. patent application Ser. No. 10/688,857, filed on Oct. 17, 2003, which claims priority to U.S. Provisional Patent Application No. 60/419,776, filed Oct. 17, 2002 and U.S. Provisional Patent Application No. 60/439,366, filed on Jan. 9, 2003, all of which are commonly assigned, and hereby incorporated by reference for all purposes.
The present invention relates generally to telecommunication techniques. More particularly, the invention provides an encoding and decoding system and method that support a plurality of compression standards and share computational resources. Merely by way of example, the invention has been applied to Code Excited Linear Prediction (CELP) techniques, but it would be recognized that the invention has a much broader range of applicability. A further example of the invention is a multi-codec that combines two or more speech or audio codecs. A wide range of speech and/or audio codecs may be integrated within the multi-codec architecture.
Code Excited Linear Prediction (CELP) speech coding techniques are widely used in mobile telephony, voice trunking and routing, and Voice-over-IP (VoTP). Such coders/decoders (codecs) model voice signals as a source filter model. The source/excitation signal is generated via adaptive and fixed codebooks, and the filter is modeled by a short-term linear predictive coder (LPC). The encoded speech is then represented by a set of parameters which specify the filter coefficients and the type of excitation.
The present invention relates generally to telecommunication techniques. More particularly, the invention provides an encoding and decoding system and method that support a plurality of compression standards and share computational resources. Merely by way of example, the invention has been applied to Code Excited Linear Prediction (CELP) techniques, but it would be recognized that the invention has a much broader range of applicability.
FIGS. 1A and 1B are simplified illustrations of the encoder and decoder modules for voice coding to encode to and decode from multiple voice coding standards;
FIG. 18 is a simplified block diagram of an embodiment of an encoder of a thin codec for SMV and EVRC according to an embodiment of the present invention;
FIG. 19 is a simplified block diagram of an embodiment of an decoder of a thin codec for SMV and EVRC according to an embodiment of the present invention;
FIG. 20 is a simplified architecture of an embodiment of a multi-codec for EFR, AMR-NB and AMR-WB encoder blocks;
FIG. 21 is a simplified architecture of an embodiment of a multi-codec for EFR, AMR-NB and AMR-WB decoder blocks;
FIG. 22 is a simplified architecture of an embodiment of a multi-codec for SMV and EVRC encoder blocks; and
FIG. 23 is a simplified architecture of an embodiment of a multi-codec for SMV and EVRC decoder blocks.
Pre-emphasis filter H(z) = 1 − 0.68z−1
W(z) = A(z/γ1)/(1 − 0.68z−1)
adaptive prefilter F(z) = 1/(1 − gp z−T)
F(z) = 1/(1 − 0.85 z−T) (1 − b1 z−1)
Separate quantization of codebook gain
and pitch gain
Multi-Codec for GSM Codecs
In one preferred embodiment, the Multi-codec architecture is applied to integrate the GSM-EFR, AMR-NB and AMR-WB speech codecs. The foundation structure of the Multi-Codec for GSM-EFR, AMR-NB and AMR-WB is the AMR-NB code. FIG. 20 shows a basic block diagram of the Multi codec encoder blocks. To integrate bit-exact EFR functionality into the Multi-Codec, additional EFR DTX/CNG functions, code and tables are added. A description of how to add AMR-WB functionality is provided herein.
The pre-processing block for AMR-NB comprises highpass filtering and downscaling. Additional functions to perform upsampling/downsampling and lowpass filtering are added, as well as the AMR-WB lowpass, highpass and tilt filter coefficients.
The LP analysis block comprises autocorrelation calculation, lag windowing and Levinson-Durbin recursion. The encoder function calling routine is adapted to activate the LP analysis routine twice per frame in the case of EFR and 12.2 kbps AMR and once per frame in all other cases. Differing input parameters are the analysis window length in samples, the table accessed for the window coefficients (which is added), and the order of prediction. LP parameter quantization for AMR-WB requires conversion of LP coefficients to ISP coefficients. Instead of adding this additional code, it can be shown that the first 15 ISPs are the same as the line spectral pairs (LSPs) derived from 15th order LP analysis, and the 16th ISP is the 16th linear prediction coefficient (LPC). Hence, the ISPs can be calculated using the AMR-NB LPC-to-LSP and LSP-to-LPC conversion functions with minor alterations. AMR-WB ISP quantization code and tables are added.
The open-loop pitch search block consists primarily of a maximum autocorrelation search on a given signal. The input signal is the weighted speech for AMR-NB, and a filtered, downsampled version of the weighted speech for AMR-WB. The lag weighting functions are identical in form, with slightly different constant values. Code additions for WB include weighting parameters, pitch range values, and interpolating filter coefficients to find � and � sample resolution for the closed-loop pitch search block. The quantization of absolute and relative pitch delays used in NB can be shared by WB.
The VAD processing block for AMR-NB comprises 2 options, the first of which is the basis for the AMR-WB VAD approach. While most of the program code is identical, some minor additions need to be made to VAD option 1 such as the AMR-WB VAD filterbank to include frequencies up to 6.4 kHz.
The ACELP codebook search block comprises computing the target signal, pre-calculation of search vectors, and testing particular pulse combinations. The fixed codebook search in AMR-WB and AMR-NB coders is one of the largest functions in terms of program size. This is due to specific fast search methods applied to reduce the number of pulse combinations tested. An exhaustive search can be compactly expressed using nested loops, however, the fast search is individual to each mode and takes up much more space to specify the order tracks are searched, the number of pulse positions optimized at once, and criteria needed to enter each stage. Further, there is a different codebook structure, number of pulses allowed and search combinations for almost every rate. The standard NB search is replaced with a unified ACELP search procedure that adapts to varying codebook structures, track orientation, pulse constraints and search conditions. The procedure has identical variable pre-calculations, specific outer layers which relate to the search order; and identical inner layers which relate to the actual combination testing. This can easily save over 50% of the reference implementation.
High band gain calculation is required for the 23.85 AMR-WB mode and must be added.
FIG. 21 shows a basic block diagram of the MULTI-CODEC decoder flow. The bitstream unwrapper block comprises decoding the bitstream back to the speech model parameters. Full ISP and fixed codebook shape decoding functions will need to be added for AMR-WB, and slight adaptations made to the NB decoding functions for the common remaining parameters. In addition, ISP extrapolation and spectrum mapping functions are required.
The excitation reconstruction and synthesis block comprises forming the excitation signal by adding the gain-scaled adaptive and fixed codebook contributions, including anti-sparseness processing, and adaptive gain control. Additional functions for noise and pitch enhancement are added for AMR-WB.
The post-processing block includes tilt compensation and formant postfiltering. For WB, function calls for highpass filtering, upsampling/downsampling, and addition of the high-band signal are required. The high-band generation block is only applicable to AMR-WB, and thus must be added in its entirety to the base codec.
No closed-loop search
guantization
Multi-Codec for CDMA Codecs
In a second preferred embodiment, the Multi-Codec architecture is applied to integrate the SMV and EVRC codecs. The foundation program code for this embodiment is the SMV program code. This is due to the large comparative size of SMV, which encompasses a broad selection of processing tools. A description of how to integrate EVRC functionality is provided herein. FIG. 22 shows a basic block diagram of the Multi-Codec for SMV and EVRC encoder blocks. The flow for SMV is not a simple direct flow as in EVRC, as it uses combined closed-loop, open-loop analysis (COLA) and repetitively loops back to recalculate and refine each parameter. Thus, in one possible implementation, separate main encoder interfaces that call shared modules are used. EVRC uses 3 subframes for all rates which equivalent to SMV Rate �, Frame Type 1 processing. Hence, the SMV code will already accommodate the appropriate subframe lengths in most cases.
The pre-processing block for SMV comprises silence enhancement, highpass filtering, noise suppression (2 options) and adaptive tilt filtering. All that is needed to be added for EVRC are the highpass filter coefficients, and a function call to cascade three SMV 2nd order filters. The EVRC noise suppression routine is identical to SMV noise suppression Option A.
The LP analysis block comprises autocorrelation calculation, lag windowing and Levinson-Durbin recursion. LP analysis is performed three times per frame in the case of SMV and once per frame for EVRC. The algorithms are identical with the exception of different analysis window lengths, analysis window coefficients, and lag window constants. These values are added, in addition to EVRC line spectral pair (LSP) quantization code and tables and large spectral transition flag calculations.
The open-loop pitch search block comprises finding the maximum autocorrelation of a given signal. The input signal is the weighted speech for SMV, and a filtered, downsampled version of the residual for EVRC. The EVRC pitch search is also follows an autocorrelation approach, but is far simpler than the SMV search, hence only small code additions are required. The closed-loop pitch search block is only applicable to SMV. The pitch lag quantization algorithm for EVRC is a subset of the quantization code already present in the SMV standard.
The rate determination block comprises functions to set the transmission rate and classify the frame type. The EVRC rate determination is identical to one of the SMV VAD options.
The RCELP signal modification block comprises forming an interpolated pitch contour and modifying the speech to match this contour. Interpolating filter coefficients and small functions to form the EVRC delay contour are added. The pulse shifting functions are shared, as SMV uses a dual warp/shift approach, part of which is the same as the pulse shifting of EVRC.
The fixed codebook search block comprises 2 main parts: ACELP codebooks and noise-excited codebooks. The fixed codebook search functions for the higher rates in SMV and EVRC coders are the largest in terms of program size. A similar approach to that described in the first embodiment can be applied here. SMV uses a more efficient different grouping and factorization of variables in calculations, which will lead to reduced EVRC complexity.
The EVRC fixed and adaptive gains are separately encoded. EVRC gain tables are added to the UTC, and corresponding gain quantization code.
FIG. 23 shows a basic block diagram of the UTC decoder flow. The bitstream unwrapper block comprises decoding the bitstream back to the speech model parameters. Full EVRC bitstream decoding functions will need to be added.
The excitation reconstruction and synthesis block comprises warping the adaptive codebook with the decoded lag, forming the excitation signal by adding the gain-scaled adaptive and fixed codebook contributions. For the post-processing block, apart from allowing for different filter coefficients and weighting factors, no code in addition to the standard SMV code is needed.
For the high performance approach, in addition to the common modules, the functions performing RCELP pulse peak picking, delay contour selection and target signal computation are modified from the standard and a common technique is applied to both standards.
Patent CitationsCited PatentFiling datePublication dateApplicantTitleUS5787390 *Dec 11, 1996Jul 28, 1998France TelecomMethod for linear predictive analysis of an audiofrequency signal, and method for coding and decoding an audiofrequency signal including application thereofUS6115688 *Aug 16, 1996Sep 5, 2000Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V.Process and device for the scalable coding of audio signalsUS6115689 *May 27, 1998Sep 5, 2000Microsoft CorporationScalable audio coder and decoderUS6167373 *Dec 30, 1999Dec 26, 2000Matsushita Electric Industrial Co., Ltd.Linear prediction coefficient analyzing apparatus for the auto-correlation function of a digital speech signalUS6314393Mar 16, 1999Nov 6, 2001Hughes Electronics CorporationParallel/pipeline VLSI architecture for a low-delay CELP coder/decoderUS6424939 *Mar 13, 1998Jul 23, 2002Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V.Method for coding an audio signalUS6717955May 7, 1999Apr 6, 2004Telefonaktiebolaget Lm Ericsson (Publ)Data communications method and apparatusUS6799060 *Sep 15, 1999Sep 28, 2004Samsung Electronics Co., Ltd.Apparatus and method for providing dialing announcement in a telephone terminalUS6807524 *Oct 27, 1999Oct 19, 2004Voiceage CorporationPerceptual weighting device and method for efficient coding of wideband signalsUS6912584 *Dec 23, 2002Jun 28, 2005Microsoft CorporationMedia coding for loss recovery with remotely predicted data unitsUS7254533 *Oct 17, 2003Aug 7, 2007Dilithium Networks Pty Ltd.Method and apparatus for a thin CELP voice codecUS7539612 *Jul 15, 2005May 26, 2009Microsoft CorporationCoding and decoding scale factor informationUS20020028670Jul 31, 2001Mar 7, 2002Nec CorporationA mobile telephone with voice data compression and recording featuresUS20030103524Sep 27, 2002Jun 5, 2003Koyo HasegawaMultimedia information providing method and apparatus* Cited by examinerNon-Patent CitationsReference13GPP TS 26.073 "ANSI-C code for the Adaptive Multi Rate (AMR) speech codec", Release 5.00, (Mar. 2002) 3rd Generation Partnership Project (3GPP), http://www.3gpp2.org/.23GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech codec; Transcoding functions", Release 5.0.0 (Jun. 2002), 3rd Generation Partnership Project (3GPP), http://www.3gpp2.org/.33GPP TS 26.104 "ANSI-C code for the floating-point AMR speech codec", Release 5.00, (Jun. 2002) 3rd Generation Partnership Project (3GPP), http://www.3gpp2.org/.43GPP TS 26.173 "ANSI-C code for the Adaptive Multi-Rate Wideband speech codec", (Mar. 2002) 3rd Generation Partnership Project (3GPP), http://www.3gpp2.org/.53GPP TS 26.190 "AMR Wideband speech codec; Transcoding Functions (Release 5)", 3rd Generation Partnership Project (3GPP); Dec. 2001, http://www.3gpp2.org/.63GPP TS 26.204 "ANSI-C code for the floating-point Adaptive Multi-Rate Wideband (AMR-WB) speech codec", Release 5.0.0, (Mar. 2002) 3rd Generation Partnership Project (3GPP); http://www.3gpp2.org/.73GPP2 C.S0030-0 "Selectable Mode Vocoder Service Option for Wideband Spread Spectrum Communication Systems", 3rd Generation Partnership Project (3GPP2) , Dec. 2001, http://www.3gpp2.org/.8ANSI/TIA/EIA-136-Rev.C, part 410-"TDMA Celluar/PCS-Radio Interface, Enhance Full Rate Voice Codec (ACELP)." Formerly IS-641. TIA published standard, Jun. 1, 2001, http://www.tiaoline.org.9ANSI/TIA/EIA-136-Rev.C, part 410�"TDMA Celluar/PCS�Radio Interface, Enhance Full Rate Voice Codec (ACELP)." Formerly IS-641. TIA published standard, Jun. 1, 2001, http://www.tiaoline.org.10Cox, "Speech Coding Standards," Speech Coding and Synthesis, W.B.Kleijn et al., eds., pp. 49-78, Elsevier Science, (1995), The Netherlands.11ETSI GSM 06.20, "Half rate speech: Half rate speech transcoding", version 8.01 (Nov. 2000). European Telecommunications Standards Institute (ETSI), http:IIwww.etsi.org/.12ETSI GSM 06.60, "Enhanced Full Rate (EFR) Speech transcoding" version 8.0.1 (Nov. 2000), European Telecommunications Standards Institute (ETSI), http:IIwww.etsi.org/.13ETSI, GSM 6.10 "Recommendation GSM 6.10 Full-Rate Speech Transcoding", version 8.02 (Nov. 2000). European Telecommunications Standards Institute (ETSI), http:IIwww.etsi.org/.14ISO/IEC 14496-3 MPEG4-CELP Coder, "Information Technology-coding of Audiovisual Objects, Part3: Audio, Subpart 3: CELP", ISO/JTC 1/SC 29 N2203CELP, May 1998.15ISO/IEC 14496-3 MPEG4-CELP Coder, "Information Technology�coding of Audiovisual Objects, Part3: Audio, Subpart 3: CELP", ISO/JTC 1/SC 29 N2203CELP, May 1998.16ITU-T G.723.1 "Speech Coders: Dual rate speech coder for multimedia communications transmission at 5.3 and 6.3 kbit/s", ITU-T Recommendation G.723.1 (1996), Geneva, http://www.itu.org/.17ITU-T G.723.1 Annex B "Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s, Annex B: Alternative specification based on floating point arithmetic", ITU-T Recommendation G.723.1-Annex B, http:IIwww.itu.org/.18ITU-T G.723.1 Annex B "Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s, Annex B: Alternative specification based on floating point arithmetic", ITU-T Recommendation G.723.1�Annex B, http:IIwww.itu.org/.19ITU-T G.728 "Coding of speech at 16 kbit/s using low-delay code excited linear prediction". ITU-Recommendation G.728 (1992), Geneva, http://www.itu.org/.20ITU-T G.728 "Coding of speech at 16 kbit/s using low-delay code excited linear prediction". ITU�Recommendation G.728 (1992), Geneva, http://www.itu.org/.21ITU-T G.729 "Coding of speech at 8 kbit/s using conjugate-Structure Algebraiccode-excited.linear-prediction (CS-ACELP)", ITU-T Recommendation G.729 (1996), Geneva, http://www.itu.org/.22ITU-T G.729A "Coding of Speech at 8 kbit/s using conjugate structure algebraiccode-excited linear-prediction (CS-ACELP) Annex A: Reduced complexity 8kbit/s CS-ACELP speech codec", ITU-T Recommendation G.729-Annex A, Nov. 1996, http://www.itu.org/.23ITU-T G.729A "Coding of Speech at 8 kbit/s using conjugate structure algebraiccode-excited linear-prediction (CS-ACELP) Annex A: Reduced complexity 8kbit/s CS-ACELP speech codec", ITU-T Recommendation G.729�Annex A, Nov. 1996, http://www.itu.org/.24ITU-T G.729C "Annex C: Reference floating-point implementation for G.729 CSACELP 8 kbit/s speech coding", ITU-T Recommendation G.729-Annex C, Sep. 1998, http://www.itu.org/.25ITU-T G.729C "Annex C: Reference floating-point implementation for G.729 CSACELP 8 kbit/s speech coding", ITU-T Recommendation G.729�Annex C, Sep. 1998, http://www.itu.org/.26Spanias, A.S. "Speech Coding: A Tutorial Review", Proc. IEEE, vol. 82, No. 10, pp. 1541-1582, Oct. 1994.27TIA/EIA/IS-127-2 "Enhanced Variable Rate Codec, Speech Service Option 3 for WidebandSpread Spectrum Digital Systems" Telecommunications Industry Association 1999.28TIA/EIAIIS-733, "High Rate Speech Service Option 17 for Wideband Spread Spectrum Communication Systems", TIA published standard, Nov. 17, 1997.Referenced byCiting PatentFiling datePublication dateApplicantTitleUS7917370 *Nov 5, 2007Mar 29, 2011National Central UniversityConfigurable common filterbank processor applicable for various audio standards and processing method thereofUS8229749 *Dec 9, 2005Jul 24, 2012Panasonic CorporationWide-band encoding device, wide-band LSP prediction device, band scalable encoding device, wide-band encoding methodUS8914280 *Jul 28, 2009Dec 16, 2014Samsung Electronics Co., Ltd.Method and apparatus for encoding/decoding speech signalUS20090292537 *Dec 9, 2005Nov 26, 2009Matsushita Electric Industrial Co., Ltd.Wide-band encoding device, wide-band lsp prediction device, band scalable encoding device, wide-band encoding methodUS20100114566 *Jul 28, 2009May 6, 2010Samsung Electronics Co., Ltd.Method and apparatus for encoding/decoding speech signalWO2013096875A2 *Dec 21, 2012Jun 27, 2013Huawei Technologies Co., Ltd.Adaptively encoding pitch lag for voiced speech* Cited by examinerClassifications U.S. Classification704/219, 704/200.1, 704/223, 704/229, 704/262, 704/220International ClassificationG10L19/04Cooperative ClassificationG10L19/16European ClassificationG10L19/16Legal EventsDateCodeEventDescriptionJun 4, 2014FPAYFee paymentYear of fee payment: 4Jun 30, 2008ASAssignmentOwner name: VENTURE LENDING & LEASING IV, INC., CALIFORNIAEffective date: 20080605Free format text: SECURITY INTEREST;ASSIGNOR:DILITHIUM NETWORKS, INC.;REEL/FRAME:021193/0242Owner name: VENTURE LENDING & LEASING V, INC., CALIFORNIAFree format text: SECURITY INTEREST;ASSIGNOR:DILITHIUM NETWORKS, INC.;REEL/FRAME:021193/0242Effective date: 20080605RotateOriginal ImageGoogle Home - Sitemap - USPTO Bulk Downloads - Privacy Policy - Terms of Service - About Google Patents - Send FeedbackData provided by IFI CLAIMS Patent Services