Patent application number | Description | Published |
20080304678 | AUDIO TIME SCALE MODIFICATION ALGORITHM FOR DYNAMIC PLAYBACK SPEED CONTROL - A modified synchronized overlap add (SOLA) algorithm for performing high-quality, low-complexity audio time scale modification (TSM) is described. The algorithm produces good output audio quality with a very low complexity and without producing additional audible distortion during dynamic change of the audio playback speed. The algorithm may achieve complexity reduction by performing the maximization of normalized cross-correlation using decimated signals. By updating the input buffer and the output buffer in a precise sequence with careful checking of the appropriate array bounds, the algorithm may also achieve seamless audio playback during dynamic speed change with a minimal requirement on memory usage. | 12-11-2008 |
20090055171 | BUZZ REDUCTION FOR LOW-COMPLEXITY FRAME ERASURE CONCEALMENT - A system is described that performs periodic waveform extrapolation based frame erasure concealment (FEC) to generate frames of an output speech signal corresponding to erased frames of encoded bit-stream in a manner reduces buzzy and tonal artifacts in the output speech signal. An embodiment of the invention uses a multiple of a pitch period associated with previously-decoded speech to perform periodic waveform extrapolation for consecutively-erased frames in a frame erasure beyond the first erased frame. An embodiment of the invention also attenuates the extrapolated signal after a threshold number of erased frames so as to reduce the FEC output signal to zero, wherein the threshold number of erased frames is dependent at least in part on the pitch period associated with the previously-decoded speech. | 02-26-2009 |
20090240492 | PACKET LOSS CONCEALMENT FOR SUB-BAND PREDICTIVE CODING BASED ON EXTRAPOLATION OF SUB-BAND AUDIO WAVEFORMS - A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame. | 09-24-2009 |
20090248405 | PACKET LOSS CONCEALMENT FOR A SUB-BAND PREDICTIVE CODER BASED ON EXTRAPOLATION OF EXCITATION WAVEFORM - Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders. | 10-01-2009 |
20090281797 | BIT ERROR CONCEALMENT FOR AUDIO CODING SYSTEMS - A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements. | 11-12-2009 |
20090281815 | COMPENSATION TECHNIQUE FOR AUDIO DECODER STATE DIVERGENCE - A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size. | 11-12-2009 |
20090282298 | BIT ERROR MANAGEMENT METHODS FOR WIRELESS AUDIO COMMUNICATION CHANNELS - Systems and methods are described for managing bit errors present in an encoded bit stream representative of a portion of an audio signal, wherein the encoded bit stream is received via a channel in a wireless communications system. The channel may comprise, for example, a Synchronous Connection-Oriented (SCO) channel or an Extended SCO (eSCO) channel in a Bluetooth wireless communications system. | 11-12-2009 |
20100125454 | PACKET LOSS CONCEALMENT FOR SUB-BAND CODECS - Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec. | 05-20-2010 |
20100192033 | VOICE ACTIVITY DETECTION (VAD) DEPENDENT RETRANSMISSION SCHEME FOR WIRELESS COMMUNICATION SYSTEMS - A voice activity detection (VAD) dependent retransmission scheme is described that mitigates the effect of packet loss on an audio signal transmitted between terminals in a wireless communication system in a manner that is generally more robust than conventional state-of-the art packet loss concealment algorithms but that consumes less terminal power as compared to conventional retransmission schemes. In one implementation, this is achieved by allowing retransmissions to be requested by a terminal only when a packet received by the terminal is deemed bad and when a portion of an audio signal currently being received by the terminal is deemed to comprise active speech. In other implementations, the processing of retransmission requests received by a terminal is inhibited or turned off entirely during periods when a portion of an audio signal currently being transmitted by the terminal is deemed not to comprise active speech. | 07-29-2010 |
20110022904 | MODEM-ASSISTED BIT ERROR CONCEALMENT FOR AUDIO COMMUNICATIONS SYSTEMS - Systems and methods are described for managing bit errors present in a series of encoded bits representative of a portion of an audio signal, wherein the series of encoded bits is received over a communication link in an audio communications system. At least one characteristic of a portion of a received modulated carrier signal that is demodulated to produce the series of encoded bits is determined. A number of bit errors present in the series of encoded bits is then determined based on the at least one characteristic. Based on the estimated number of bit errors, one of a plurality of methods for producing a series of digital audio samples representative of the portion of the audio signal is selectively performed. The series of digital audio samples produced by the selected method is then converted into a form suitable for playback to a user. | 01-27-2011 |
20110029317 | DYNAMIC TIME SCALE MODIFICATION FOR REDUCED BIT RATE AUDIO CODING - Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder. | 02-03-2011 |
20110099008 | BIT ERROR MANAGEMENT AND MITIGATION FOR SUB-BAND CODING - Systems and method for managing and/or mitigating the impact of bit errors on encoded frames received by an LC-SBC (Low Complexity Sub-band Coding) decoder are described herein. For example, in one embodiment, the impact of bit errors on an LC-SBC frame received by an LC-SBC decoder is estimated and one of a plurality of bit error management techniques is applied to the LC-SBC frame based on the estimated impact, wherein the bit error management techniques may include performing PLC, performing normal SBC decoding, or performing some other technique for managing and/or mitigating the impact of the bit errors. Techniques for concealing bit errors in LC-SBC frames are also described. | 04-28-2011 |
20110099009 | NETWORK/PEER ASSISTED SPEECH CODING - A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals. | 04-28-2011 |
20110099014 | SPEECH CONTENT BASED PACKET LOSS CONCEALMENT - Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal. | 04-28-2011 |
20110099015 | USER ATTRIBUTE DERIVATION AND UPDATE FOR NETWORK/PEER ASSISTED SPEECH CODING - Systems, methods and apparatuses are described for deriving and updating user attribute information about users of a communications system. A communications network is then used to transfer the user attribute information to communication terminals, which use the user attribute information to configure a speech codec to operate in a speaker-dependent manner during a communication session, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals. | 04-28-2011 |
20110099019 | USER ATTRIBUTE DISTRIBUTION FOR NETWORK/PEER ASSISTED SPEECH CODING - Systems, methods and apparatuses are described herein for distributing user attribute information about users of a communications system to communication terminals, which use the user attribute information to configure a speech codec to operate in a speaker-dependent manner during a communication session, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals. | 04-28-2011 |
20110208517 | TIME-WARPING OF AUDIO SIGNALS FOR PACKET LOSS CONCEALMENT - Packet loss concealment (PLC) systems and methods are described that use time-warping to merge a concealment signal generated to replace one or more bad frames of an audio signal with a received signal representing one or more subsequent good frames of the audio signal in a manner that avoids signal discontinuity and audible artifacts resulting therefrom. Prediction-based PLC systems and methods are also described that use time-warping to conceal the loss of one or more frames containing a transition region in a manner that will not result in an audible artifact. | 08-25-2011 |
20110209029 | Low complexity error correction using cyclic redundancy check (CRC) - Low complexity error correction using cyclic redundancy check (CRC). Communications between at communication devices, sometimes including at least one redundant transmission from a transmitter to a receiver, undergo low complexity error correction. CRC may be employed in conjunction with using any desired type of ECC or using uncoded modulation. Based on CRC determined bit-errors, as few as a singular syndrome associated with a singular bit-error or a linear combination of syndromes associated with two or more singular bit-errors within two or more received signal sequences are employed to perform error correction of the received signal. Real time combinations of multiple syndromes associated with respective single bit-errors (that may themselves be calculated off-line) are employed in accordance with error correction. In addition to CRC, any ECC may be employed including convolutional code, RS code, turbo code, TCM code, TTCM code, LDPC code, or BCH code. | 08-25-2011 |
20110320213 | TIME-WARPING OF DECODED AUDIO SIGNAL AFTER PACKET LOSS - A technique is described for use in a decoder configured to decode a series of frames representing an encoded audio signal. The technique is for transitioning between a lost frame and one or more received frames following the lost frame in the series of frames. In accordance with the technique, an output audio signal associated with the lost frame is synthesized. An extrapolated signal is generated based on the synthesized output audio signal. A time lag is calculated between the extrapolated signal and a decoded audio signal associated with the received frame(s), wherein the time lag represents a phase difference between the extrapolated signal and the decoded audio signal. The decoded audio signal is time-warped based on the time lag, wherein time-warping the decoded audio signal comprises stretching or shrinking the decoded audio signal in the time domain. | 12-29-2011 |
20120010882 | CONSTRAINED AND CONTROLLED DECODING AFTER PACKET LOSS - A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal. | 01-12-2012 |
20130179161 | NETWORK/PEER ASSISTED SPEECH CODING - A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals. | 07-11-2013 |
20130188758 | JOINT SOURCE CHANNEL DECODING USING PARAMETER DOMAIN CORRELATION - Methods, systems, and apparatuses are provided for performing joint source channel decoding in a manner that exploits parameter domain correlation. Redundancy in speech coding and packet field parameters is exploited to generate conditional probabilities that a decoder utilizes to perform joint source channel decoding. The conditional probabilities are based upon correlations of parameters of a current frame with parameters of the same or other frames or historical parameter data. Parameter domain correlation provides significant channel decoding improvement over prior bit domain solutions. Also provided are methods, systems, and apparatuses for utilizing received statistics of monitored data bits from which conditional probabilities are generated to perform channel decoding. The techniques described may be implemented at the decoder side and thus do not interfere with transmission standards. | 07-25-2013 |
20130191120 | CONSTRAINED SOFT DECISION PACKET LOSS CONCEALMENT - Methods, systems, and apparatuses for performing packet loss concealment are disclosed. In response to determining that an encoded frame representing a segment of a signal is bad, an encoded parameter within the encoded frame is decoded based on bit information (such as soft bit information) associated with the encoded parameter to obtain a decoded parameter. Whether the decoded parameter violates a parameter constraint is determined. If a parameter constraint violation is detected, an estimate of the decoded parameter is generated. Either the decoded parameter or estimate of the decoded parameter is passed to a decoder for use in decoding the encoded frame. | 07-25-2013 |
20130191706 | MODEM ARCHITECTURE FOR JOINT SOURCE CHANNEL DECODING - A modem architecture that supports the application of joint source channel decoding (JSCD). The modem architecture includes two channel decoders, one of which is modified to provide improved signal quality. The modem architecture further includes transparent network layers that enable the passage of data from one layer to another layer. For example, the modem architecture enables the passage soft bits, when available, from a physical layer to an application layer. The soft bits may be utilized for JSCD, packet loss concealment, or other applications. The modem architecture enables encryption and decryption of data to incorporate extrinsic information in operating JSCD. | 07-25-2013 |
20130191707 | MODIFIED JOINT SOURCE CHANNEL DECODER - A turbo decoder is configured to perform iterative decoding of data bits of a data packet received via a source signal to converge on a “soft” decision representation for each data bit of the data packet. The turbo decoder includes both an interleaved decoder and a non-interleaved decoder that work collaboratively to refine and improve the “soft” decision of each of the originally-received data bits. The interleaved decoder and the non-interleaved decoder are injected with extrinsic information based on at least a-priori information of the source signal. The turbo decoder avoids positive feedback of the a-priori information regarding the source signal from one decoder to the other by subtracting out extrinsic information based on the a-priori information that is injected into a decoder from the “soft” decision(s) determined by the decoder. | 07-25-2013 |
20130282366 | JITTER BUFFER ENHANCED JOINT SOURCE CHANNEL DECODING - Methods, systems, and apparatuses are provided for performing jitter buffer enhanced joint source channel decoding. Jitter buffer enhanced joint source channel decoding may be performed in a manner that exploits parameter domain correlation. A jitter buffer stores hard bits of properly channel decoded packets, and a secondary jitter buffer is implemented to store soft bits associated with packets that are improperly channel decoded. Joint source channel decoding may be delayed to perform channel decoding of a frame in the penultimate position of the jitter buffer. The soft bits stored in the secondary jitter buffer as well as hard bits stored in the jitter buffer, which may include future frames, are utilized to perform channel decoding. The delayed jitter buffer enhanced joint source channel decoding may also be extended to iteratively perform channel decoding for giving frames at each position in the jitter buffer as they traverse the jitter buffer. | 10-24-2013 |
20130346826 | Low complexity error correction using cyclic redundancy check (CRC) - Low complexity error correction using cyclic redundancy check (CRC). Communications between at communication devices, sometimes including at least one redundant transmission from a transmitter to a receiver, undergo low complexity error correction. CRC may be employed in conjunction with using any desired type of ECC or using uncoded modulation. Based on CRC determined bit-errors, as few as a singular syndrome associated with a singular bit-error or a linear combination of syndromes associated with two or more singular bit-errors within two or more received signal sequences are employed to perform error correction of the received signal. Real time combinations of multiple syndromes associated with respective single bit-errors (that may themselves be calculated off-line) are employed in accordance with error correction. In addition to CRC, any ECC may be employed including convolutional code, RS code, turbo code, TCM code, TTCM code, LDPC code, or BCH code. | 12-26-2013 |
20140032227 | BIT ERROR MANAGEMENT METHODS FOR WIRELESS AUDIO COMMUNICATION CHANNELS - Systems and methods are described for managing bit errors present in an encoded bit stream representative of a portion of an audio signal, wherein the encoded bit stream is received via a channel in a wireless communications system. The channel may comprise, for example, a Synchronous Connection-Oriented (SCO) channel or an Extended SCO (eSCO) channel in a Bluetooth® wireless communications system. | 01-30-2014 |
20140059407 | CHASE CODING FOR ERROR CORRECTION OF ENCRYPTED PACKETS WITH PARITY - A plurality of encrypted packets having common payload data are received, wherein each of the plurality of encrypted packets includes a corresponding parity check field, and wherein a corresponding parity check syndrome for each of the plurality of encrypted packets indicates at least one bit error. A payload portion of each of the plurality of encrypted packets is decrypted to generate a plurality of decrypted payload portions. At least one chase coding technique is used to generate a corrected decrypted payload, based on at least one candidate bit error position and further based on the corresponding parity check syndrome for at least one of the plurality of encrypted packets. | 02-27-2014 |
20140278397 | SPEAKER-IDENTIFICATION-ASSISTED UPLINK SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device. | 09-18-2014 |
20140278417 | SPEAKER-IDENTIFICATION-ASSISTED SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device. | 09-18-2014 |
20140278418 | SPEAKER-IDENTIFICATION-ASSISTED DOWNLINK SPEECH PROCESSING SYSTEMS AND METHODS - Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in a downlink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the far-end speaker is then used to improve the performance of one or more downlink speech processing algorithms implemented on the communication device. | 09-18-2014 |