Abstract: A system of specifying link layer information in a URL is described. In an embodiment, a URL is generated which includes both a link layer network type and information which is used by a resolving device to identify a particular link layer network of the specified type. In various embodiments, the URL includes a link layer network type and a corresponding link layer network name or pairs of link layer network types and corresponding link layer network names. Where the URL comprises more than one link layer network name, the resolving device may determine at runtime which of the named link layer networks to connect to and this decision may be based on criteria or preference information included within the URL.

Abstract: A system includes a controller that provides an output control signal based on two or more control inputs. The controller determines an indication of radio frequency (RF) bandwidth availability based on a given one of the control inputs. The output control signal can correspond to the RF bandwidth availability and status of at least one other wireless condition affecting RF bandwidth. An audio quality adjuster can adjust quality parameters used to encode an audio stream for one or more audio sinks based on the output control signal.

Abstract: A method is disclosed for maintaining spatial queues in digital sound signals. Sound signals are received from each of a plurality of transducers. The sound signals are transformed using a common real-valued spectral gain, G, to maintain spatial cues within the sound signals, the common spectral gain, G, determined by: calculating G as a function of a derivative of a known cost function and as a function of at least one multichannel frequency-domain Bayesian short-time estimator.

Abstract: A noise injection system adds comfort noise to an audio signal. The system includes a background noise estimator that determines a spectral content of a background noise associated with the audio signal. A comfort noise generator generates a comfort noise signal having a random phase. A gain circuit adjusts the comfort noise signal based on the spectral content of the background noise. A combining circuit combines a gain-adjusted comfort noise signal and the audio signal to generate an output signal.

Abstract: Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.

Abstract: An apparatus for processing an input sound signal, the apparatus including: gain circuitry configured to control a gain based on a plurality of respective sub-signals of the input sound signal; and an amplification apparatus configured to adjust the amplification of all the plurality of amplitudes based on the gain.

Abstract: The voice quality optimization system includes a controller that controls voice quality by adjusting parameters that control voice quality characteristics of the communication device; and a measuring unit that measures voice quality of the communication device and transmits the measured voice quality as a feedback to the controller. The controller controls voice quality by calibrating the parameters of the communication device, including a receiving sensitivity/frequency response characteristic curve, receiving loudness rating and idle channel noise-receiving. A method for setting voice optimization in a communication device includes measuring parameters of the communication device, determining whether the parameters of the communication device are within a target range, and calibrating a first parameter to be within the target range if the first parameter is outside the target range.

Abstract: Disclosed is a coding apparatus and method using residual bits. Accordingly, performance (voice quality) is enhanced by quantizing a full-band gain of frequency coefficients existing in sub-bands to which bits are not assigned in an algebraic vector quantization (AVQ). Further, the performance (voice quality) is enhanced by sequentially quantizing a sub-band gain of sub-bands to which bits are not assigned until residual bits are removed. Furthermore, the performance (voice quality) is enhanced by demodulating AVQ coefficients, and correcting quantization noises starting with a coefficient having the greatest absolute coefficient among the AVQ coefficients, when residual bits additionally remain.

Abstract: A speech enhancement method is disclosed. The method includes the steps of: receiving a plurality of frames of sound signals by a microphone array; calculating an inter-aural time difference for each frequency band of each frame of the sound signals corresponding to at least one two-microphone set of the microphone array; calculating a plurality of values of cumulative histograms according to the calculated inter-aural time differences; determining a first inter-aural time difference threshold according to the calculated value of the cumulative histograms; and filtering the plurality of frames of sound signals according to the first inter-aural time difference threshold.

Abstract: This invention provides a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.

Abstract: A speech signal is received at an input. At least one electrical value associated with the received speech signal is tracked. A dynamic adjustment of the speech signal is determined. The dynamic adjustment is selected at least in part so as to minimize a distortion and minimize an over-amplification of the speech signal based at least in part upon an analysis of the at least one electrical value. The dynamic adjustment is further selected to obtain a desired output signal characteristic for the speech signal presented at an output. The dynamic adjustment value is applied to the speech signal and the adjusted speech signal is presented at the output. The gain of the signal can also be limited to prevent over-amplification.

Abstract: A signal manipulator for manipulating an audio signal having a transient event may have a transient remover, a signal processor and a signal inserter for inserting a time portion in a processed audio signal at a signal location where the transient event was removed before processing by the transient remover, so that a manipulated audio signal has a transient event not influenced by the processing, whereby the vertical coherence of the transient event is maintained instead of any processing performed in the signal processor, which would destroy the vertical coherence of a transient.

Abstract: According to one embodiment, a speech processing apparatus includes a histogram calculation unit, a cumulative frequency calculation unit, and a filter production unit. The histogram calculation unit is configured to calculate a first histogram from a first speech feature extracted from speech data, and to calculate a second histogram from a second speech feature different from the first speech feature. The cumulative frequency calculation unit is configured to calculate a first cumulative frequency by accumulating a frequency of the first histogram, and to calculate a second cumulative frequency by accumulating a frequency of the second histogram. The filter production unit is configured to produce a filter having a characteristic to get the second cumulative frequency near to the first cumulative frequency.

Abstract: Described herein are methods, systems, apparatuses and products for reconstruction of a smooth speech signal from a stuttered speech signal. One aspect provides for accessing a stored speech signal having stuttering; identifying at least one stuttered region in the stored speech signal; modifying the at least one stuttered region in the stored speech signal; and responsive to modifying the at least one stuttered region, reconstructing a smooth speech signal corresponding to the stored speech signal. Other embodiments are disclosed.

Abstract: A method, user device and computer program product for processing audio signals during a communication session between a user device and a remote node. The method comprising: receiving a plurality of audio signals at audio input means at the user device including at least one primary audio signal and unwanted signals; receiving direction of arrival information of the audio signals at a gain control means; providing to the gain control means known direction of arrival information representative of at least some of said unwanted signals; processing the audio signals at the gain control means by applying a level of gain to generate a gain controlled signal for transmission to the remote node, wherein the level of gain applied is dependent on a comparison between the direction of arrival information of the audio signals and the known direction of arrival information.

Abstract: Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame.

Abstract: A speech signal processing system that includes a speech input unit for inputting a speech signal; input speech storage unit for storing an input speech signal that is the speech signal inputted through the speech input unit; characteristic estimation unit for referring to the input speech signal stored in the input speech storage unit, and estimating characteristics of an input speech indicated by the input speech signal, the characteristics including an environmental sound included in the input speech signal; reference speech output unit for causing a predetermined speech signal that becomes a reference speech, to output; and characteristic adding unit for adding the characteristics of the input speech estimated by the characteristic estimation unit, in a reference speech signal that is the speech signal caused to output by the reference speech output unit.

Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

Abstract: In an audio output terminal device, a buffer control unit adjusts the buffer size of a jitter buffer in accordance with the setting of a sound output mode instructed in an instruction receiving unit. If the instruction receiving unit acknowledges an instruction for setting an audio output mode that requires low delay in outputting sound, the buffer control unit reduces the buffer size of the jitter buffer. Further, the buffer control unit controls, in accordance with the instructed setting of the sound output mode, timing for allowing a media buffer to transmit one or more voice packets to the jitter buffer.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: Disclosed are systems, methods, and computer readable media for performing speech recognition. The method embodiment comprises selecting a codebook from a plurality of codebooks with a minimal acoustic distance to a received speech sample, the plurality of codebooks generated by a process of (a) computing a vocal tract length for a each of a plurality of speakers, (b) for each of the plurality of speakers, clustering speech vectors, and (c) creating a codebook for each speaker, the codebook containing entries for the respective speaker's vocal tract length, speech vectors, and an optional vector weight for each speech vector, (2) applying the respective vocal tract length associated with the selected codebook to normalize the received speech sample for use in speech recognition, and (3) recognizing the received speech sample based on the respective vocal tract length associated with the selected codebook.

Abstract: An apparatus is provided with a device storing machine readable code and a processor executing the machine readable code. The machine readable code includes sound setting code and audio processing code. The sound setting code detects use of a microphone and sets sound characteristics that are suitable for conversation in response to detecting the use of the microphone. The audio processing code processes sound on the basis of the sound characteristics set by the sound setting code.

Abstract: A noise injection system adds comfort noise to an audio signal. The system includes a background noise estimator that determines a spectral content of a background noise associated with the audio signal. A comfort noise generator generates a comfort noise signal having a random phase. A gain circuit adjusts the comfort noise signal based on the spectral content of the background noise. A combining circuit combines a gain-adjusted comfort noise signal and the audio signal to generate an output signal.

Abstract: A system and method for automatically adjusting floor controls based on conversational characteristics is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold comprising a minimum number of timeslices for at least one of the current configuration and one of the possible configurations is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: A voice reproduction apparatus includes an ambient sound analysis unit to analyze a characteristic of an ambient sound, a characteristic analysis unit to analyze an acoustic characteristic of a signal for reproduction, a reproduction timing adjusting unit to record the signal for reproduction and to read the signal for reproduction at a reproduction timing of follow-up reproduction, a reproduction speed changing unit to change a reproduction speed of the read signal for reproduction, and a control unit to control the reproduction timing adjusting unit so that the signal for reproduction is reproduced at the reproduction timing corresponding to an analysis result of the ambient sound analysis unit and to control the reproduction speed changing unit so that the signal for reproduction is reproduced at the reproduction speed corresponding to the analysis result of the ambient sound analysis unit and the acoustic characteristic obtained by the characteristic analysis unit.

Abstract: A warped spectral estimate of an original audio signal can be used to encode a representation of a fine estimate of the original signal. The representation of the warped spectral estimate and the representation of the fine estimate can be sent to a speech recognition system. The representation of the warped spectral estimate can be passed to a speech recognition engine, where it may be used for speech recognition. The representation of the warped spectral estimate can also be used along with the representation of the fine estimate to reconstruct a representation of the original audio signal.

Abstract: A speech processing apparatus and method. The speech processing apparatus includes a microphone to receive a speech signal, an analog/digital converter to convert the speech signal generated by the microphone into a digital speech signal, and an automatic gain controller to calculate an average value of the magnitude of the digital speech signal generated by the analog/digital converter in a plurality of frames, to determine in which region of a speech signal band the average value is located, the speech signal band being divided into a plurality of regions according to the strength of speech, and to adjust gain according to a location of the average value on the speech signal band so that the strength of speech has a level of an optimal region capable of processing the speech signal. Accordingly, speech recognition may be maximized without being constrained by the distance of a speech source.

Abstract: Provided is a technology which adjusts an input signal such that the volume of a conversation or speech contained in a content is substantially constant, thereby alleviating the audience from a burden of making a volume control operation. An acoustic signal processor comprises an acoustic signal storage unit which buffers an acoustic input signal for a predetermined period of time; a voice detection unit which detects a voice section from the buffered acoustic signal; an acoustic signal-to-loudness level transformation which calculates a loudness level from the buffered acoustic signal; a threshold/level comparator which compares the calculated loudness level with a predetermined target level; a voice amplification calculation unit which calculates a gain control amount for the buffered acoustic signal on the basis of the detection and comparison results; and an acoustic signal amplifier which amplifies or dampens the buffered acoustic signal in accordance with the calculated gain control amount.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

Abstract: An apparatus may include a control unit to selectively control volume of content sound and volume of speech sound according to a priority assigned to a user corresponding to speech sound and a priority assigned to content data. When volume control is to be performed on a priority basis, the control unit may selectively control the volume of the content sound and the volume of the speech sound based on the assigned priorities so that the volume of the sound having a higher priority becomes louder than the volume of the other sound.

Abstract: Provided is a communication device which can easily provide a function to enable signal cross-reference among a plurality of voice signals having different frequency ranges and to enhance the quality of voice communications, at a low cost. In the communication device, a band expansion unit (104) expands a narrow band voice signal to a broad frequency range voice signal; a gain control unit (105) controls a gain of a downstream voice signal using an upstream voice signal as a reference signal; an echo canceler (109) cancels an echo component contained in the upstream voice signal using the downstream voice signal as a reference signal; a down-sampling unit (111) generates a narrow band reference signal by performing a band conversion to convert a broad band voice signal to a narrow band voice signal; and an up-sampling unit (112) generates a broadband reference signal by performing band conversion to convert a narrow band voice signal to a broadband voice signal.

Abstract: An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.

Abstract: A voice processing apparatus includes a voice signal acquiring unit that acquires a voice signal converted to plural frequency bands from an input signal having a narrowed band; an expanding unit that generates based on a narrowband component of the voice signal acquired by the voice signal acquiring unit, an expansion band component expanding the band of the voice signal; a correcting unit that corrects the power of the expansion band component by a correction amount determined based on a noise component included in the voice signal acquired by the voice signal acquiring unit; and an output unit that outputs the voice signal of which the band has been expanded based on the expansion band component corrected by the correcting unit and based on the narrowband component of the voice signal acquired by the voice signal acquiring unit.

Abstract: When a frame immediately preceding an encoding target frame to be encoded by a first encoding unit operating under a linear predictive coding scheme is encoded by a second encoding unit operating under a coding scheme different from the linear predictive coding scheme, the encoding target frame can be encoded under the linear predictive coding scheme by initializing the internal state of the first encoding unit. Therefore, encoding processing performed under a plurality of coding schemes including the linear predictive coding scheme and a coding scheme different from the linear predictive coding scheme can be realized.

Abstract: A speech signal processing system which outputs a speech feature, divides an input speech signal into frames so that each pair of consecutive frames have a frame shift length equal to at least one period of the speech signal and have an overlap equal to at least a predetermined length, applies discrete Fourier transform to each of the frames, calculates a CSP coefficient for the pair, searches a predetermined search range in which a speech wave lags a period equal to at least one period to obtain the maximum value of the CSP coefficient for the pair, and generates time-series data of the maximum CSP coefficient values arranged in the order in which the frames appear. A method and a computer readable article of manufacture for the implementing the same are also provided.

Abstract: A method for performing a call between a near-end user and a far-end user, which includes the following operations performed during the call by the near-end user's communications device. Automatic gain control (AGC) is performed to update a gain applied to an uplink speech signal. A frame is detected in a downlink signal that contains speech; in response, the updating of the gain is frozen. Other embodiments are also described and claimed.

Abstract: Methods of audio coding are described in which an excitation signal for a first frequency band of the audio signal is used to calculate an excitation signal for a second frequency band of the audio signal that is separated from the first frequency band.

Abstract: Method and decoder for processing of audio signals. The method and decoder relate to deriving a processed vector {circumflex over (d)} by applying a post-filter directly on a vector d comprising quantized MDCT domain coefficients of a time segment of an audio signal. The post-filter is configured to have a transfer function H which is a compressed version of the envelope of the vector d. A signal waveform is reconstructed by performing an inverse MDCT transform on the processed vector {circumflex over (d)}.

Abstract: A portable terminal and method for adjusting transmitted sound output to a wireless headset. A portable terminal includes a wireless connection setting unit to establish a connection with a wireless headset and a loopback control unit. The loopback control unit may transmit a test signal within a voice band to the connected wireless headset, receive a sound signal inputted through a microphone of the portable terminal in response to the transmitted test signal, analyze the received sound signal and adjust sound output to be transmitted to the wireless headset. A method for adjusting sound output includes establishing a connection with a wireless headset, transmitting a test signal within a voice band, receiving a sound signal inputted through a microphone of the portable terminal in response to the transmitted test signal, analyzing the sound signal, and adjusting sound output to be transmitted to the wireless headset according to the analysis result.

Abstract: A signal processing method is provided. The signal processing method includes extracting a first signal having a first frequency band from a sum signal of a left signal and a right signal, generating a second signal having a second frequency band by using the first signal, generating a third signal by using the first signal and the second signal, and applying a gain, generated by using a rate of a center signal included in the sum signal, to the third signal.

Abstract: A system capable of reducing the influence of sound reverberation or reflection to improve sound-source separation accuracy. An original signal X(?,f) is separated from an observed signal Y(?,f) according to a first model and a second model to extract an unknown signal E(?,f). According to the first model, the original signal X(?,f) of the current frame f is represented as a combined signal of known signals S(?,f?m+1) (m=1 to M) that span a certain number M of current and previous frames. This enables extraction of the unknown signal E(?,f) without changing the window length while reducing the influence of reverberation or reflection of the known signal S(?,f) on the observed signal Y(?,f).

Abstract: A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold.

Abstract: The present invention relates to the field of speech recognition and synthesis, and more specifically to a novel non-phonemically based system for recognizing and synthesizing speech.

Abstract: Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.

Abstract: Provided are an apparatus and a method for integrally encoding and decoding a speech signal and a audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is a audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: In one embodiment the present invention includes a method of improving audibility of speech in a multi-channel audio signal. The method includes comparing a first characteristic and a second characteristic of the multi-channel audio signal to generate an attenuation factor. The first characteristic corresponds to a first channel of the multi-channel audio signal that contains speech and non-speech audio, and the second characteristic corresponds to a second channel of the multi-channel audio signal that contains predominantly non-speech audio. The method further includes adjusting the attenuation factor according to a speech likelihood value to generate an adjusted attenuation factor. The method further includes attenuating the second channel using the adjusted attenuation factor.

Abstract: A method and apparatus for encoding and decoding a residual signal are provided. The encoding method includes generating a residual signal indicating a difference between a multi-channel audio signal, and an audio signal downmixed from the multi-channel audio signal and then upmixed by using additional information from the downmixed audio signal; and performing a parametric encoding method on the residual signal. The decoding method includes decoding a sinusoidal component; restoring a sine wave by using the sinusoidal component; dividing the sine wave into a plurality of sub-bands in a frequency domain; transforming the plurality of sub-bands from the frequency domain into a time domain by applying a window to each of the plurality of sub-bands; and synthesizing the plurality of domain-transformed sub-bands to restore a residual signal.

Abstract: Good sound quality as perceived by the ear is obtained even with few information bits. A shape quantizer (111) is comprised of an interval search unit (121) which searches and encodes the pulses in each band of a plurality of divisions of the specified search interval, and a full search unit (122) which searches for pulses over the entire search interval, and quantizes the shape of the input spectrum at the positions and the polarities of a small number of pulses. The interval search unit (121) encodes a pulse searched for in a higher band than the specified frequency with fewer bits than a pulse searched for in another band. The full search unit (122) encodes the pulses positioned in a higher band than the specified frequency with fewer bits than the other pulses. A gain quantizer (112) calculates and quantizes in each band the gain of a pulse searched for by the shaper quantizer (111).

Abstract: The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.