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 |
20090006084 | LOW-COMPLEXITY FRAME ERASURE CONCEALMENT - A system is described that performs frame erasure concealment (FEC) to generate frames of an output speech signal corresponding to erased frames of encoded bit-stream in a manner that conceals the quality-degrading effects of such erased frames. An embodiment of the invention advantageously does not introduce additional delay, has lower state memory requirement than the FEC technique specified in G.711 Appendix I, and produces better speech quality than the FEC technique specified in G.711 Appendix I while still allowing for reduced computational complexity and code size. | 01-01-2009 |
20090111507 | Speech intelligibility in telephones with multiple microphones - The present invention is directed to improved speech intelligibility in telephones with multiple microphones. Such a telephone includes a first microphone, a second microphone, a voice activity detector (VAD), a receiver module, and a signal processor. The first microphone outputs a first audio signal, which comprises a voice component when a near-end user talks and a background noise component. The second microphone outputs a second audio signal. The VAD generates a voice activity signal responsive to a ratio between the first audio signal and the second audio signal. The voice activity signal identifies time intervals in which the voice component of the near-end user is present in the first audio signal. The receiver module receives a third audio signal, which comprises a voice component of a far-end user. The signal processor modifies the third audio signal responsive to the voice activity signal. | 04-30-2009 |
20090209290 | Wireless Telephone Having Multiple Microphones - The present invention is directed to a wireless telephone having a first microphone and a second microphone and a method for processing audio signal in a wireless telephone having a first microphone and a second microphone. The wireless telephone includes a first microphone, a second microphone, and a signal processor. The first microphone outputs a first audio signal, the first audio signal comprising a voice component and a background noise component. The second microphone outputs a second audio signal. The signal processor increases a ratio of the voice component to the noise component of the first audio signal based on the content of at least one of the first audio signal and the second audio signal to produce a third audio signal. | 08-20-2009 |
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 |
20090281800 | SPECTRAL SHAPING FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20090281801 | COMPRESSION FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20090281802 | SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND METHOD - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20090281803 | DISPERSION FILTERING FOR SPEECH INTELLIGIBILITY ENHANCEMENT - A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal. | 11-12-2009 |
20090281805 | INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER - A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith. | 11-12-2009 |
20090287496 | LOUDNESS ENHANCEMENT SYSTEM AND METHOD - A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs. | 11-19-2009 |
20110029304 | HYBRID INSTANTANEOUS/DIFFERENTIAL PITCH PERIOD CODING - A hybrid instantaneous/differential encoding technique is described herein that may be used to reduce the bit rate required to encode a pitch period associated with a segment of a speech signal in a manner that will result in relatively little or no degradation of a decoded speech signal generated using the encoded pitch period. The hybrid instantaneous/differential encoding technique is advantageously applicable to any speech codec that encodes a pitch period associated with a segment of a speech signal. | 02-03-2011 |
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 |
20110255713 | AUTOMATIC VOLUME CONTROL FOR AUDIO SIGNALS - A technique is provided for automatically adjusting the volume, or magnitude, of an audio signal. The technique includes calculating an average power associated with a segment of an input audio signal, determining whether the average power is greater than an estimated signal level associated with one or more previously-processed segments of the input audio signal and, depending on the determination, either calculating an updated estimated signal level by subtracting from the average power an attenuated difference between the estimated signal level and the average power or setting the updated estimated signal level to the average power. A gain to be applied to the segment of the input audio signal is then determined based on the updated estimated signal level and a target signal level for an output audio signal. | 10-20-2011 |
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 |
20120101824 | PITCH-BASED PRE-FILTERING AND POST-FILTERING FOR COMPRESSION OF AUDIO SIGNALS - Systems and methods for enhancing the quality of an audio signal produced by an audio codec are described herein. In accordance with the systems and methods, a pitch-based pre-filter adaptively filters an input audio signal to produce a filtered audio signal. An audio encoder encodes the filtered audio signal to generate a compressed audio bit stream. An audio decoder decodes the compressed audio bit stream to generate a decoded audio signal. A pitch-based post-filter adaptively filters the decoded audio signal to produce an output audio signal, wherein adaptively filtering the decoded audio signal comprises undoing at least part of a signal-shaping effect of the pitch-based pre-filter. | 04-26-2012 |
20120121100 | Method and Apparatus For Wind Noise Detection and Suppression Using Multiple Microphones - Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described. | 05-17-2012 |
20120123771 | Method and Apparatus For Wind Noise Detection and Suppression Using Multiple Microphones - Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described. | 05-17-2012 |
20120123772 | System and Method for Multi-Channel Noise Suppression Based on Closed-Form Solutions and Estimation of Time-Varying Complex Statistics - Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system. | 05-17-2012 |
20120123773 | System and Method for Multi-Channel Noise Suppression - Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage. | 05-17-2012 |
20130142037 | ADAPTIVE PACKET SIZE MODIFICATION FOR PACKET NETWORKS - A system and method for modifying the size of data packets transmitted over a packet network in a manner that avoids overloading of the network. The system and method involves monitoring one or more parameters indicative of an amount of bandwidth being utilized on the packet network, responsive to the monitoring, determining that a level of bandwidth utilization on the packet network has changed, responsive to the determination that the level of bandwidth utilization on the packet network has changed, issuing a command to change the size of packets used for carrying data from a first packet size to a second packet size. | 06-06-2013 |
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 |
20130195163 | SYSTEMS AND METHODS FOR ENHANCING AUDIO QUALITY OF FM RECEIVERS - Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L−R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L−R noise signal. A channel quality measure is calculated based on the L−R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L−R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L−R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation. | 08-01-2013 |
20130195164 | SYSTEMS AND METHODS FOR ENHANCING AUDIO QUALITY OF FM RECEIVERS - Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, a stop band noise signal is extracted from an L+R or L−R signal produced by an FM stereo decoder. A channel quality measure is calculated based on the stop band noise signal and is used to control whether a pop suppression technique is applied to the L+R signal. The channel quality measure and the stop band noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on the L−R signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation. | 08-01-2013 |
20130216057 | ECHO CANCELLATION USING CLOSED-FORM SOLUTIONS - A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal. | 08-22-2013 |
20130346072 | NOISE FEEDBACK CODING FOR DELTA MODULATION AND OTHER CODECS - Systems and methods are described that apply a noise feedback coding (NFC) technique at the encoder of a delta modulation codec, such as a Continuously Variable Slope Delta Modulation (CVSD) codec, so as to shape the spectrum of the coding noise produced thereby in such a way that the speech quality of the delta modulation decoder output is enhanced. The techniques described herein are not limited to delta modulation codecs and may also be applied to any sample-by-sample codec, including a G.711 μ-law codec, a linear pulse code modulation (LPCM), or any other of a wide variety of sample-by-sample codecs, to improve the audio quality of the decoder output thereof. | 12-26-2013 |
20140188466 | INTEGRATED SPEECH INTELLIGIBILITY ENHANCEMENT SYSTEM AND ACOUSTIC ECHO CANCELLER - A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith. | 07-03-2014 |
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 |
20140286497 | MULTI-MICROPHONE SOURCE TRACKING AND NOISE SUPPRESSION - Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described. | 09-25-2014 |