Abstract: The invention concerns sound spatialization with multichannel encoding for binaural reproduction on two loudspeakers, the spatial encoding being defined by encoding functions associated with multiple encoding channels and the decoding by applying filters for binaural reproduction.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: An analog input stage has m differential input channels, wherein m>1. The analog input stage is configured to select one of the m differential input channels and provide an output signal. The analog input stage has n identical selection units each having m differential channel inputs and one differential output, wherein n is at least 2m?1. Each selection unit is operable to be coupled to any of the differential input channels through respective differential multiplexer units, wherein the multiplexor units are driven to select one of the differential input channels and couple the selected differential channel input through a butterfly switch unit with the differential output of the selection unit. The differential output signals of the n selection units are combined whereby unwanted crosstalk from channels other than a selected channel are removed by cancellation.

Abstract: A method of enhancing vertical polar localization of a head related transfer (HRTF). The method includes splitting an audio signal and generating left and right output signals by determining a log lateral component of the respective frequency-dependent audio gain that is equal to a median log frequency-dependent audio gain for all audio signals of that channel having a desired perceived source location. A vertical magnitude of the respective audio signal is enhanced by determining a log vertical component of the respective frequency-dependent audio gain that is equal to a product of a first enhancement factor and a different between the respective frequency-dependent audio gain at the desired perceived source location and the lateral magnitude of respective audio signal. The output signals are time delayed according to an interaural time.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: A system for simultaneous visual data presentation is provided having a processor and memory with instructions for execution by the processor for: receiving attended data; receiving unattended data; and generating presentation data. The presentation is suitable for simultaneous presentation of the attended data and the unattended data in a manner in which the unattended data at least partially overlaps the attended data and the unattended data does not interfere with the attended data by replacing, erasing, or suppressing the attended data, and vice versa. The presented unattended data gives cues that the user can sense and provide information to the user while attending to the presented attended data, without attending to the presented unattended data.

Abstract: An encoding device, a decoding device, and encoding and decoding methods are provided, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.

Abstract: An audio or video enhancement module incorporates advanced audio enhancement effects. These advanced effects cover a wide range including but not limited to 3D audio spatialization (virtualization) and transient enhancement. The device is configured for receiving an audio input signal, in analog or digital form, enhancing the received signal, and transmitting an output audio signal having the enhanced effects embedded in the signal. The output signal may be either digital or audio. Further, the output signal may be a mono signal, a stereo signal, a MIDI signal, or a multichannel signal (such as 2.1, 5.1, 7.1, etc.). User input to the module is provided to control the amount of audio enhancement.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for stereophonic acoustic echo cancellation. The method includes collecting, at a same time, a first audio sample of an audio source from a first omnidirectional microphone and a second audio sample of the audio source from a second omnidirectional microphone. The method includes delaying the second audio sample by a first amount of time to yield a delayed second audio sample and combining the delayed second audio sample with the first audio sample to produce a first channel, then delaying the first audio sample by a second amount of time to yield a delayed first audio sample and combining the delayed first audio sample with the second audio sample to produce a second channel. Then the method includes outputting the first channel and the second channel as a stereo audio signal of the audio source.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: An audio system for a vehicle includes a directional loudspeaker that is mounted to a vehicle seat. The directional loudspeaker radiates a first channel audio signal so that the direction toward an intended location of a first ear position of an occupant of the vehicle seat is a high radiation direction and radiates a second channel audio signal so that the direction toward an intended location of a second ear position of the occupant of the vehicle seat is a high radiation direction. A forward mounted loudspeaker radiates at least one of the first channel audio signal and the second channel audio signal. Signal processing circuitry modifies (e.g., delays) the first channel audio signal to at least one of the directional loudspeaker and the forward mounted loudspeaker to cause one of those speakers to dominate spatial perception.

Abstract: Embodiments are described for a method and system of rendering and playing back spatial audio content using a channel-based format. Spatial audio content that is played back through legacy channel-based equipment is transformed into the appropriate channel-based format resulting in the loss of certain positional information within the audio objects and positional metadata comprising the spatial audio content. To retain this information for use in spatial audio equipment even after the audio content is rendered as channel-based audio, certain metadata generated by the spatial audio processor is incorporated into the channel-based data. The channel-based audio can then be sent to a channel-based audio decoder or a spatial audio decoder. The spatial audio decoder processes the metadata to recover at least some positional information that was lost during the down-mix operation by upmixing the channel-based audio content back to the spatial audio content for optimal playback in a spatial audio environment.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: Methods and systems for processing an audio signal are provided. The method includes generating a pseudorandom sequence and generating at least one reciprocal of the pseudorandom sequence such that the at least one reciprocal is substantially decorrelated with the pseudorandom sequence. The pseudorandom sequence and the at least one reciprocal form a set of sequences. The method further includes convolving the audio signal with the set of sequences to generate a corresponding number of output signals and providing the number of output signals to a corresponding number of loudspeakers.

Abstract: An audio signal is processed in the frequency domain to convert an input signal format to an output signal format. That is, a multichannel audio signal intended for playback over a predefined speaker layout can be formatted to achieve spatial reproduction over a different layout comprising a different number of speakers.

Abstract: An apparatus comprising: a first audio signal analyser configured to analyse a first audio signal to determine at least one audio source, wherein the at least one audio source has a virtual location; a second audio signal analyser configured to analyse a second audio signal to determine at least one localised audio source, wherein the second audio signal is generated from the apparatus audio environment; and a repositioner configured to reposition the virtual location of the at least one audio source dependent on the at least one localised audio source.

Abstract: Signal processing section of a terminal converts acquired audio signals of a plurality of channels into frequency spectra set, calculates sound image positions corresponding to individual frequency components, and displays, on a display screen, the calculated sound image positions results by use of a coordinate system having coordinate axes of the frequency components and sound image positions. User-designated partial region of the coordinate system is set as a designated region and an amplitude-level adjusting amount is set for the designated region, so that the signal processing section adjusts amplitude levels of frequency components included in the frequency spectra and in the designated region, converts the adjusted frequency components into audio signals and outputs the converted audio signals.

Abstract: A first speaker radiates a first sound toward first and second listening points by a first sound signal. A second speaker radiates a second sound toward these points by a second sound signal. The first sound signal is generated by convolving a signal with a first filter coefficient. The second sound signal is generated by convolving the signal with a second filter coefficient. A nearer speaker to a virtual sound source is decided among the first and second speakers. The first and second filter coefficients are calculated so that a ratio of a first sound pressure of the first and second sounds at the first listening point to a second sound pressure thereof at the second listening point is equal to a target ratio. Among the first and second filter coefficients, based on one filter coefficient corresponding to the nearer speaker, the other filter coefficient corresponding to another speaker is calculated.

Abstract: Techniques for personalizing audio levels, in accordance with embodiments of the present technology, provide different audio volumes to different locations in a room allowing for two of more users to enjoy the same audio content at different volumes. Differential level and delay compensation filtering based on a position of each of a plurality of speakers, the location of each user and the preferred relative audio volume of each user are utilized to produce different effective audio levels in localized regions of a room.

Abstract: Disclosed are an active delay method and an improved wireless binaural hearing device. The binaural hearing device includes: a first hearing device including a first microphone, an amplifier and a wireless transmitter; and a second hearing device including a second microphone, an amplifier, a wireless transmitter, a wireless receiver which receives a signal from the wireless transmitter, an active delay circuit which synchronizes the received signal with a signal acquired by the second microphone, a neural network which synchronizes the delayed signal, and a speaker which converts the synchronized signal into a voice signal. With this configuration, it is possible to prevent incorrect detection of the position of the sound source or paralalia due to a time delay which is produced in the wireless binaural hearing device and reduce noises due to a time difference between both hearing devices, thereby providing a binaural hearing device with high quality.

Abstract: A method and apparatus are provided for generating a personalized radio channel playlist by simultaneously buffering tracks from multiple received channels from one or more source streams and selectively playing back tracks from the buffered channels. Navigation tools permit users to skip buffered songs in their playlist (e.g., skip forward and backward). Users can specify favorite channels for building personal playlists, or multiple default playlist channels can be provided (e.g., by genre). Thumbs up/down buttons on the radio receiver permit entering a song or artist being played back into a favorites list that is used to search all channels for matches or a banned list used to block songs from future playlists. A matched channel carrying the favorite can be added to a playlist. Segments on the playlist can be played back in full or truncated to facilitate preview of playlist segments.

Abstract: A sound reproduction apparatus comprising a loudspeaker bar having a plurality of loudspeakers. A loudspeaker bar controller coupled to the loudspeaker bar for processing audio data for the plurality of loudspeakers, the loudspeaker bar controller comprising a spatial enhancement/virtualization system for receiving a surround channel of audio data and processing the surround channel of audio data with a spatial generation/virtualization filter, wherein a left stereo channel of audio data and a right stereo channel of audio data are not processed with the head related transfer function filter. Bass is enhanced for small speakers which are not able to produce bass frequencies.

Abstract: Multiple independent sound images are formed by integrally performing uncorrelation processing and sound image localization processing on an input audio signal with signal processors that use output functions hl(x) and hr(x) obtained by integrating an uncorrelation function that generate multiple audio signals with low mutual correlation from an input audio signal. The signal processors have a sound image localization function for localizing the sound image of each of the multiple audio signals at a given sound source position.

Abstract: An apparatus, method, and medium of removing crosstalk are provided. The apparatus for removing crosstalk in each of audio signals of a plurality of channels, includes: a position recognition unit recognizing the position of a listener; a calculation unit calculating a crosstalk removal function with respect to the recognized position; and a filtering unit removing the crosstalk by using the calculated result.

Abstract: An audio enhancement technique comprising an audio source, processing the audio source by the following modules, a low pass filter with dynamic offset, an envelope controlled bandpass filter, a high pass filter; and adding an amount of dynamic synthesized sub bass to the audio. The processed audio signals are combined in a summing mixer with the audio source to create an audio out signal. Audio out is sent to a block that splits, processes and combines the stereo stream into several different versions that are fed into the Stereo Bus A. The Stereo Bus is fed to a Compare Block that adjusts the amplitude of the original and processed audio by averaging.

Abstract: A method for producing a digital spatialized stereo audio file from an original multichannel audio file, comprising a step of performing a processing on each of the channels for cross-talk cancelation; a step of merging the channels in order to produce a stereo signal; and a dynamic filtering and specific equalization step for increasing the sound dynamics.

Abstract: A method of navigating in a sound content wherein at least one key word is stored in association with at least two positions representative of said key word in the sound content, and wherein the method comprises: a step of displaying a representation of the sound content; during playback of the sound content, a step of detecting a current extract representative of a key word stored at a first position; a step of determining at least one second extract representative of said key word and a second position as a function of the stored positions; and a step of highlighting the position of the extracts in the representation of the sound content. The invention also relates to a system adapted to implement the navigation method.

Abstract: A method and apparatus of expanding a listening sweet spot. A method of expanding a listening sweet spot with respect to signals output from speakers includes: obtaining an HRTF (head related transfer function) at a position of a listener's ear; moving a first virtual ear around the position of the listener's ear; obtaining an HRTF at each position of the first virtual ear; and processing a signal to be input to the speakers using the obtained HRTFs to output to the speakers.

Abstract: The invention discloses a method and an encoder for processing a digital audio stereo signal. A digital audio encoder for coding such audio signal comprises a predictive Temporal Noise Shaping (TNS) filter, a Mid-/Side (M/S) coding unit, a control unit for determining a first prediction gain related to the unmodified L/R signal processed by the TNS filter and for determining a second prediction gain related to the M/S-coded L/R signal processed by the TNS filter, wherein the control unit is adapted to disable TNS-filtering—i.e. to bypass the TNS filter—for a current signal frame, if the first and second prediction gains differ by more than a pre-determined mismatch range.

Abstract: The present invention relates to an acoustic signal processing apparatus that suppresses auditory noise caused in a difference signal generated by acoustic signals of a plurality of channels, a processing method therefor, and a program. A difference spectrum calculation unit 320 calculates, as a difference spectrum, the absolute difference value between frequency spectra of the left and right channels sent from frequency spectrum generation units 311 and 312. A low-level band determination unit 330 determines a difference spectrum corresponding to a low-level band among difference spectra of all the frequency bands. A substitution spectrum generation unit 350 generates a substitution spectrum for which a difference spectrum is substituted, on the basis of the frequency spectrum of the left channel. A spectrum substitution unit 360 substitutes the difference spectrum corresponding to the low-level band for a substitution spectrum corresponding to the difference spectrum.

Abstract: A home use sound reproduction system for Hi-fi, with digital signal transfer from a playback unit via a control unit to one or more active loudspeakers, each including or arranged beside an amplifier unit. The control unit is arranged to control sound parameters and send both a digital sound information signal and a power signal for powering the amplifier units, and to superimpose the single ended or differential digital signal together with the power signal on at least one common lead in a cable.

Abstract: A system and method for generating virtual microphone signals having a particular number and configuration for channel playback from an intermediate set of signals that were recorded in an initial format that is different from the channel playback format. In one embodiment, an initial set of intermediate are Bark-banded such that each intermediate signal may lead to a corresponding power spectral density (PSD) signal representative of the initial intermediate signal. Further, one may generate cross-correlations signals for each pair of intermediate signals. Next, from the PSDs and cross correlations, one may more efficiently calculate corresponding channel signals to be used for playback on respective channel speakers. Thus, the PSDs of each channel signal may be generated at chosen angles (as well as other design factors). Further, each channel signal may also be further modified with a corresponding cancellation signal that further enhances the resultant signal in each channel.

Abstract: The invention generally relates to acoustic transducers. In certain aspects, the acoustic transducer includes a diaphragm and an actuator. In further aspects, the acoustic transducer includes a connector having a planar proximal end that tapers to a distal end. The proximal end may be coupled to the diaphragm and the distal end coupled to the actuator, such that the actuator causes movement of the diaphragm. In additional aspects, the acoustic transducer includes a connector having a proximal end that couples to the diaphragm and a distal end that couples to the actuator. The proximal end of the connector can include a damping element. In other aspects, the diaphragm includes a connector, in which the connector tapers to a distal end that is configured to mate with the actuator such that the actuator causes movement of the diaphragm.

Abstract: An audio signal processing device includes a processing unit for convoluting head-related transfer functions with audio signals of a plurality of channels, and the processing unit includes a storage unit for storing data of a double-normalized head-related transfer function by normalizing a normalized head-related transfer function obtained by normalizing a head-related transfer function in a state in which a dummy head or a person is present in a position of the listener with a transfer characteristic in a pristine state in which the dummy head or the person is not present, using a normalized head-related transfer function obtained by normalizing a head-related transfer function in the state in which the dummy head or the person is present with a transfer characteristic in the pristine state, and a convolution unit for reading the data from the storage unit and convoluting the data with the audio signals.

Abstract: This document discusses apparatus and methods for configuring and providing crosstalk cancellation to maintain channel separation in a multi channel system. In an example, an amplifier circuit can include a crosstalk cancellation circuit configured to reduce crosstalk from a first output to a second load and from a second output to a first load where the first load and the second load share a return path.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: In one embodiment, a sound field is mapped by extracting spatial angle information, diffusivity information, and optionally, sound level information. The extracted information is mapped for representation in the form of a Riemann sphere, wherein spatial angle varies longitudinally, diffusivity varies latitudinally, and level varies radially along the sphere. A more generalized mapping employs mapping the spatial angle and diffusivity information onto a representative region exhibiting variations in direction of arrival that correspond to the extracted spatial information and variations in distance that correspond to the extracted diffusivity information.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: A device (1) for converting a first number (M) of input audio channels into a second, larger number (N) of output audio channels comprises: decorrelation units (3) for decomposing the input audio channels into a set of decorrelated auxiliary channels, at least one upmix unit (4) for combining the decorrelated auxiliary channels into the output audio channels, and at least one pre-processing unit (2) for pre-processing the input audio channels and feeding the pre-processed input audio channels to the decorrelation units (3). The pre-processing unit (2) and the upmix unit (4) are preferably controlled by audio parameters.

Abstract: A method includes detecting, via a first microphone coupled to a user's left ear, a sound, detecting, via a second microphone coupled to the user's right ear, the sound, determining a time difference between detection of the sound at the first microphone and detection of the sound at the second microphone, and estimating a user's head size based on the time difference. The method also includes identifying a head-related transfer function (HRTF) associated with the user's head size or modifying, a HRTF based on the user's head size. The method further includes applying the identified HRTF or modified HRTF to audio signals to produce output signals and forwarding the output signals to first and second speakers coupled to the user's left and right ears.

Abstract: This document discusses, among other things, apparatus and methods including an analog-to-digital controller (ADC) configured to receive an enable signal and to provide an ADC output signal to control logic, wherein the control logic is configured to provide a control voltage to a control input of a switch. In an example, the control voltage includes the ADC output signal when the ADC output signal is below a first threshold or above a second threshold. In certain examples, the control logic is configured to transition the control voltage from the first threshold to the second threshold when the ADC output signal is between the first and second thresholds.

Abstract: An audio content playback method for a portable terminal The audio content playback method includes checking a channel that is supportable by audio content that is currently engaged in group's simultaneous playback, in group's simultaneous playback of the audio content. The method includes allocating a channel to each of devices included in a group based on position information of each device included in the group or based on an input state in a user interface environment that is preset for channel allocation for each device included in the group, and transmitting the allocated channel information to each device included in the group to allow the device to select its allocated channel and play the audio content.

Abstract: When a correlation between an L channel and an R channel of a reproduction-target sound source is considerably high, virtual sound obtained from a reproduction signal is often more likely to be localized inside the head of a listener. A device includes: a correlation analysis unit (3) that analyzes a degree of correlation between a surround L channel signal (SL signal) and a surround R channel signal (SR signal); and an output signal control unit 4 that controls, based on the degree of correlation between the SL signal and the SR signal obtained as a result of the analysis performed by the correlation analysis unit (3), a ratio between: the signals outputted from a front L speaker (7) and a front R speaker (8); and the signals outputted from a near-ear L speaker (9) and a near-ear R speaker (10).

Abstract: Disclosed is an audio signal decoding device and a method of balance adjustment that reduces a fluctuation of a decoded signal orientation and maintains a stereo perception. An interchannel correlation computation unit (224) computes a correlation between a left channel decoded stereo signal and a right channel decoded stereo signal, and if the interchannel correlation is low, a peak detection unit (225) uses a peak component of a decoded monaural signal of the current frame and a peak component of either a left or a right channel of the preceding frame to detect a peak component with a high temporal correlation. The peak detection unit (225) combines and outputs, from among the frequencies of the detected peak components, a peak frequency of a frame n?1 and a peak frequency of a frame n. A peak balance coefficient computation unit (226) computes, from the peak frequency of the frame n?1, a balance parameter that is used in converting a peak frequency component of the monaural signal to stereo.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: The present invention provides a mobile electronic device and an audio playing means thereof. The audio playing means comprises a first audio player and a second audio player for playing a left channel audio and a right channel audio respectively; a first audio receiver and a second audio receiver for receiving a first echo of the left channel audio and a second echo of the right channel audio in real time respectively; and a control unit for compensating the left channel audio and the right channel audio according to the first echo and the second echo which are received, and controlling the first audio player and the second audio player to play a compensated left channel audio and a compensated right channel audio respectively, wherein the compensated left channel audio and the compensated right channel audio are able to offset the first echo and the second echo respectively.

Abstract: Embodiments of the subject invention relate to a method and apparatus for virtualizing an audio file. The virtualized audio file can be presented to a user via, for example, ear-speakers or headphones, such that the user experiences a change in the user's perception of where the sound is coming from and/or 3D sound. Embodiments can utilize virtualization processing that is based on head rotated transfer functions (HRTF's) or other processing techniques that can alter where the user perceives the sounds of the music file to originate from. A specified embodiment provides Surround Sound virtualization with DTS Surround Sensations software. Embodiments can utilize the 2-channel audio transmitted to the headphones.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting an arbitrary down-mix signal and compensation information necessary for compensating for the arbitrary down-mix signal from the input bitstream, compensating for the arbitrary down-mix signal using the compensation information, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the compensated arbitrary down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: A system and method for automatically changing a state of a device coupled to a headphone device is provided. The system comprises a means for detecting if at least one earpiece of the headphone device is activated or deactivated. Based on an activation state or a change in an activation state of the at least one earpiece, a state change may occur in the device. The state change is determined by a preset of a control module.

Abstract: A crosstalk cancellation system and method is disclosed for use in a multi-channel communication system. Crosstalk which couples between channels is cancelled through use of an in-line FFE filter and in-line delay. A cross-connect system associated with each channel includes a cross-connect delay and cross-connect filter. The cross-connect system generates a cancellation signal for each of the channels, which is routed into a junction. The junction subtracts the cancellation signal from the received signal, which has also been delayed and filtered, to remove unwanted cross-talk. During training, cancellation magnitude is monitored at various delay offsets to determine which offset and corresponding filter coefficients, for each delay, maximizes cancellation. The filters are set with filter coefficients that maximize cancellation. The cross-connect delay with the maximum offset is set to zero and its calculated offset amount is established as the in-line delay offset.