Abstract: A system and methodology of delivery fluids and monitoring their status which is voice actuated. This system has application where a hands-free environment is preferred. Voice commands are given by the user via a Bluetooth® headset and received typically by the user's Smartphone. Voice recognition circuitry is programmed to recognize the simple commands and through complementing electronics, and electro-mechanical and mechanical elements, delivery at corresponding flow rates is accomplished. A further feature allows for respective voice commands to initiate a monitoring function where the status of any particular characteristic of the fluid can be relayed back to the user via the headset.

Abstract: According to one embodiment, a learning apparatus includes a first storage and a processing circuitry. The first storage stores a plurality of similarity information items, each including at least a second text, intention candidates and certainty degrees representing a certainty with which the intention candidates are regarded as being identical to intentions of the second text. The processing circuitry detects corresponding similarity information item including the second text identical to a first text from the plurality of similarity information items, corrects a utterance intention to an intention candidate included in the corresponding similarity information item where a certainty degree included in the corresponding similarity information item is not less than a threshold.

Abstract: In a speech recognition method, a criteria value is determined to determine the length of a silent section included in a processing section, and a processing mode to use is determined in accordance with the criteria value. The criteria value is used to obtain audio information of the processing section. Audio processing is executed on the audio information in the processing section, using the processing mode that has been determined. Speech recognition processing is executed on the audio information in the processing section that has been subjected to audio processing.

Abstract: A portable acoustic unit is adapted for insertion into an electrical receptacle. The portable acoustic unit has an integrated microphone and a wireless network interface to an automation controller. The portable acoustic unit detects spoken voice commands from users in the vicinity of the electrical receptacle. The portable acoustic unit merely plugs into a conventional electrical outlet to provide an extremely simple means of voice control through a home or business.

Abstract: Techniques related to key phrase detection for applications such as wake on voice are discussed. Such techniques may include updating a start state based rejection model and a key phrase model based on scores of sub-phonetic units from an acoustic model to generate a rejection likelihood score and a key phrase likelihood score and determining whether received audio input is associated with a predetermined key phrase based on the rejection likelihood score and the key phrase likelihood score.

Abstract: A computer detects a connected voice or video call between participants and records a brief media sample. Speech recognition is utilized to determine when the call is connected as well as to transcribe the content of the audio portion of the media sample. The recorded media sample and transcribed content is associated with the connected voice or video call such that a user may reference it at a later point. The computer additionally suggests creating or editing contact information associated with the participants of the connected voice or video call based on the transcribed content.

Abstract: An improved gain-shape vector quantization is achieved by determining a number of bits to be allocated to a gain adjustment- and shape-quantizer for a plurality of combinations of a current bit rate and a first signal property. The bit allocation is derived by using an average of optimal bit allocations for a training data set. A number of bits to the gain adjustment and the shape quantizers for a plurality of combinations of the bit rate and a first signal are pre-calculated, and a table indicating the number of bits to be allocated to the gain adjustment- and the shape-quantizers for a plurality of combinations of the bit rate and a first signal property is created. In this way, the table can be used for achieving an improved bit allocation.

Abstract: A device includes an encoder. The encoder is configured to receive two audio channels. The encoder is also configured to determine a mismatch value indicative of an amount of a temporal mismatch between the two audio channels. The encoder is further configured to determine, based on the mismatch value, at least one of a target channel or a reference channel. The target channel corresponds to a lagging audio channel of the two audio channels and the reference channel corresponds to a leading audio channel of the two audio channels. The encoder is also configured to generate a modified target channel by adjusting the target channel based on the offset value. The encoder is further configured to generate at least one encoded channel based on the reference channel and the modified target channel.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component (CAMB(k?1)). The ambient HOA component (C˜AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Methods and apparatus for audio watermarking are disclosed in which watermark data is codified in a plurality of Fourier transform coefficients of the audio signal. The watermarked audio is transmitted and captured as sound waves after analogic conversion, typically through a medium with some degree of signal degradation. The receiving end converts the watermarked audio back to the digital domain before extracting the watermark data from the Fourier transform coefficients. This configuration is enhanced in certain embodiments by a robust bit codification technique with fast decoding algorithms, synchronization signalling and error correction.

Abstract: A signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. An envelope information generating unit generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. The high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream.

Abstract: A method for separating an audio signal into a harmonic signal component and a transient signal component is disclosed. The method includes the steps of: transferring the audio signal into a frequency space in order to obtain a transferred audio signal in dependence on frequency and time and applying a non-linear smoothing filter to the transferred audio signal over frequency to obtain a filtered transient signal in which the harmonic signal component is suppressed relative to the transient signal component. The method further includes applying the non-linear smoothing filter to the transferred audio signal over time to obtain a filtered harmonic signal in which the transient signal component is suppressed relative to the harmonic signal component and determining the harmonic signal component and the transient signal component based on the filtered harmonic signal and the filtered transient signal.

Abstract: According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

Abstract: It is inter alia disclosed to determine a first quantized representation of an input vector, and to determine a second quantized representation of the input vector based on a codebook depending on the first quantized representation

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: A method of performing gain adjustment in an electronic device includes determining a first set of spectral frequency values and determining a second set of spectral frequency values. The first set of spectral frequency values corresponds to a high-band portion of an audio signal received at the electronic device. The second set of spectral frequency values approximates the first set of spectral frequency values in the high band portion of the audio signal. The method includes estimating a spectral distortion corresponding to a difference between the first set of spectral frequency values and the second set of spectral frequency values and adjusting, based on the spectral distortion, a gain value corresponding to at least a portion of the audio signal. The method also includes transmitting an encoded bitstream that includes information corresponding to the adjusted gain value and the second set of spectral frequency values.

Abstract: A method and apparatus are provided for updating post-processing states applied to a decoded audio signal. The method is such that, for a current decoded signal frame, sampled at a different sampling frequency from the preceding frame, it includes the following acts: obtaining a past decoded signal, stored for the preceding frame; re-sampling by interpolation of the past decoded signal obtained; using the re-sampled past decoded signal as a memory for post-processing the current frame. A decoding method is also provided, which includes updating post-processing states.

Abstract: An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

Abstract: A method and apparatus for enhancing audio processing between a transmit radio (230) and a receive radio (240) are provided. Digitized audio frames (202) are applied to parallel inputs of both a vocoder encoder (204) and a trill encoder 212 of the transmit radio (230). The vocoder encoder (204) generates voice bits which are communicated over a voice bits channel (206) to a vocoder decoder (208) of the receive radio (240). Trill encoder (212) generates signaling bits which are communicated over a signaling bits channel (214) to a trill decoder (216) of the receive radio (240) for recovery of trill information (218). At the receive radio (240), a decoded audio signal (209) generated from the vocoder decoder (208), and the recovered trill information (218) are both provided as inputs to a trill reconstructor stage (220) to generate a recovered audio signal (222) having a reconstructed trill.

Abstract: Embodiments of the present invention provide systems, methods, and computer storage media directed towards automatic emphasis of spoken words. In one embodiment, a process may begin by identifying, within an audio recording, a word that is to be emphasized. Once identified, contextual and lexical information relating to the emphasized word can be extracted from the audio recording. This contextual and lexical information can be utilized in conjunction with a predictive model to determine a set of emphasis parameters for the identified word. These emphasis parameters can then be applied to the identified word to cause the word to be emphasized. Other embodiments may be described and/or claimed.

Abstract: Devices, systems and methods are disclosed for reducing noise in input data by performing a hysteresis operation followed by a lateral excitation smoothing operation. For example, an audio signal may be represented as a sequence of feature vectors. A row of the sequence of feature vectors may, for example, be associated with the same harmonic of the audio signal at different points in time. To determine portions of the row that correspond to the harmonic being present, the system may compare an amplitude to a low threshold and a high threshold and select a series of data points that are above the low threshold and include at least one data point above the high threshold. The system may iteratively perform a spreading technique, spreading a center value of a center data point in a kernel to neighboring data points in the kernel, to further reduce noise.

Abstract: A method for reducing noise to a user to enable a conversation-of-interest to be heard, the noise originating from a noise source, the method comprising the steps of: operating at least one first device located at a first distance from the noise source, the user having noise-cancellation earphones connected to a second mobile device, the second mobile device located at a second distance from the noise source, the first distance less than the second distance; prehearing noise from the noise source using the at least one first device; analyzing the preheard noise to yield a respective analyzed noise signal; and processing the respective analyzed noise signal to effect noise cancellation for the noise-cancellation earphones.

Abstract: A system, method, and apparatus for detecting drones are disclosed. An example method includes receiving a digital sound sample and partitioning the digital sound sample into segments. The method also includes applying a frequency and power spectral density transformation to each of the segments to produce respective sample vectors. For each of the sample vectors, the example method determines a combination of drone sound signatures and background sound signatures that most closely match the sample vector. The method further includes determining, for the sample vectors, if the drone sound signatures in relation to the background sound signatures that are included within the respective combinations are indicative of a drone. Conditioned on determining that the drone sound signatures are indicative of a drone, an alert message indicative of the drone is transmitted.

Abstract: Embodiments of the present invention relate to detecting irregularities in audio, such as music. An input signal corresponding to an audio stream is received. The input signal is transformed from a time domain into a frequency domain to generate a plurality of frames that each comprises frequency information for a portion of the input signal. An irregular event in a portion of the input signal corresponding to a set of frames in the plurality of frames is identified based on a comparison of frequency information of the set of frames to the frequency information of other sets of frames of the plurality of frames. This allows an indication of the irregular event to be provided, or for the input signal to be automatically synchronized to a multimedia event.

Abstract: A reception unit receives a speech signal from another terminal device. A reproduction unit reproduces the speech signal received in the reception unit. A processing unit performs a speech recognition process on the speech signal reproduced in the reception unit, based on a speech recognition model of a user using the terminal device. A transmission unit transmits a result of the speech recognition process in the processing unit to another terminal device.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: A magnetic recording system for preventing data loss resulting magnetic oscillator current. The magnetic recording system includes a magnetic write head with a magnetic write pole, a magnetic oscillator near the magnetic write pole, and a write coil for magnetizing the write pole. Circuitry is connected with the magnetic write coil to supply a current to the write coil and connected with the magnetic oscillator to supply a current to the magnetic oscillator. The circuitry is configured to ensure that the current to the magnetic oscillator does not inadvertently magnetize the write pole after the magnetic write pole has demagnetized.

Abstract: A hard disk drive write drive method and integrated circuit that provide a configurable overshoot current based on the length of the pattern in the write current signal that is being written to the hard disk by the write driver. The hard disk write driver adds a first overshoot current to all patterns in the write current signal and adds an additional second overshoot current to patterns in the write current signal shorter than a first duration. The hard disk drive write driver utilizes an H-bridge circuit configured to add the first overshoot current and the additional second overshoot current to the write current signal. The H-bridge circuit is comprised of four switching elements that are configured to generate a first overshoot current for all pattern transitions of the write current signal and generate a second overshoot current for pattern transitions shorter than the first duration.

Abstract: A PMR writer is disclosed wherein magnetic flux return from a magnetic medium to a main pole is substantially greater through a trailing shield structure than through a leading shield and return path layer (RTP). Magnetic impedance is increased between the RTP and main pole in the leading return loop by modifying the size and shape of the back gap connection (BGC), by decreasing Bs in the RTP or reducing its thickness, or by removing one or more layers in the BGC and replacing with dielectric material or non-magnetic metal to form a dielectric gap between the RTP and main pole. As a result, area density control and bit error rate are improved over a conventional dual write shield (DWS) structure comprising two flux return pathways. Moreover, adjacent track erasure is maintained at a level similar to a DWS design.

Abstract: A magnetic field-assisted magnetic recording (MAMR) head is provided, which includes a recording main pole and a texture control layer (TCL), a seed control layer, and a spin torque oscillator (STO) positioned over the main pole, in this order, in a stacking direction from a leading side to a trailing side of the recording head. The STO has a crystallographic preferred growth orientation and includes a spin polarized layer (SPL). The TCL may include a Cu layer.

Abstract: A slider of a magnetic recording head has a leading edge, a trailing edge, and an air bearing surface between the leading and trailing edges. A sensor is situated at the trailing edge of the slider and configured to sense presence of a lube droplet or other contaminant at the trailing edge.

Abstract: A computer program product according to one embodiment includes a computer readable storage medium having program instructions embodied therewith, the program instructions executable by a controller of a data storage device to cause the controller to perform a method including, determining, by the controller, a test skew gain of a coil using skew information from a servo subsystem of the controller; comparing, by the controller, the test skew gain to a stored reference gain for determining whether a temperature of the coil is elevated; and stopping the data storage device, by the controller, in response to the comparing of the test skew gain to the reference gain indicating that the temperature of the coil is elevated.

Abstract: A magnetic media having a lateral exchange control layer formed on a magnetic oxide layer of a magnetic recording layer. A cap layer is formed over the lateral exchange control layer. The lateral exchange control layer can be an alloy comprising Co and one or more of W, Ru, Hf, Ta, Nb and Fe. The lateral exchange control layer has the highest magnetic saturation moment among all the recording layers, and increases spacing between magnetic grains (e.g. increased non-magnetic boundary width), thereby reducing lateral exchange sigma. The presence of lateral exchange control increases signal to noise ratio and reduces bit error rate and increases areal density.

Abstract: An optical disk drive and a digital servo method for the optical disk drive includes controlling functions of the optical disk drive with a microprocessor. Low-pass filtered and gain adjusted versions of individual photodetector output signals resulting from an illumination of an optical disk are received. Versions of the individual photodetector output signals are digitized to produce digital signals. A focus control signal and/or a tracking control signal is determined through at least one servo algorithm executed by a digital signal processor based on a focus error and/or tracking error, respectively, determined from the digital signals. The digital signal processor is an integrated circuit structured and arranged for manipulation of digital signals in accordance with programmed commands, and in a manner that operates faster than the microprocessor.

Abstract: A storage system is proposed which comprises a data storage device including a number N of removable storage media for storing data, with N?2, and a number m of drives, with m?1, wherein each of the m drives is configured to drive one of the removable storage media mounted to the drive. The storage system includes a server being coupled to the data storage device and configured to serve requests from clients, and a controller which is configured to control the data storage device to adaptively unmount the removable storage media based on a monitoring information of the requests at the data storage device, states of the drives, and states of the removable storage media.

Abstract: Authentication of discs occurs by the use of interferometric authentication data. Such authentication data of a data disc is generated based on an interference pattern associated with thickness variations of a material layer on the disc. The interference pattern represents a unique fingerprint that can be used for authentication of individual discs.

Abstract: An integrated circuit connectable to a signal transmission path of a storage device includes an impedance setting circuit including a plurality of switches, a detector configured to detect an input signal level, and a decoder. The decoder controls one or more switches of the impedance setting circuit based on the detected input signal level so as to match an impedance of the integrated circuit with an impedance of the signal transmission path.

Abstract: An apparatus includes a storage medium operable to store a number of data tracks, a read channel circuit operable to process the data tracks read from the storage medium, and a track quality classifier circuit operable to determine a track quality metric for the data tracks read from the storage medium. The track quality metric indicates whether a corresponding one of the data tracks that has failed to successfully process in the read channel circuit can be reprocessed within a track gap period.

Abstract: It is possible to accurately and securely execute conversion from an MPEG-2TS format to an MP4 format. A data converting unit that executes format conversion from an MPEG-2TS format to an MP4 format sorts out TS packets storing video data from MPEG-2TS format data, selects a TS packet having a start code (SC) at the top of a payload on the basis of header information of the sorted out TS packets, and extracts payload configuration data other than the start code (SC) and sets the payload configuration data as configuration data of MP4 format data. The data converting unit determines, on the basis of, for example, a value of a transport priority set in a header of the TS packet, whether or not the TS packet is a TS packet in which a start code (SC) is set in a top region of a payload.

Abstract: A storage device includes a controller that directs incoming data to a storage location based on a capacity of a region or surface of a magnetic disc. According to one implementation, the storage device controller writes new data to data tracks in a first series of data tracks on the magnetic disc until a capacity condition is satisfied. Once the capacity condition is satisfied, the storage device controller writes new data to a second series of data tracks on the storage medium that are interlaced with data tracks of the first series.

Abstract: According to one embodiment, when an access destination of a reading request is a second region, a controller performs a first retry reading process by use of a second reading condition in first parameter information. The second region is adjacent to a first region, which is adjacent to a defect region and has a particular width, and is present as a region other than the defect region on a magnetic disk. When the access destination of the reading request is the first region, the controller performs a second retry reading process by use of a fourth reading condition in second parameter information. The second parameter information is set such that a grown defect is registered at an earlier time in the second retry reading process, as compared with that in the first retry reading process.

Abstract: In one embodiment, a method includes reading, using a plurality of read sensors of a magnetic head in a read channel of a tape drive, encoded data from a plurality of tracks of a magnetic tape medium simultaneously. Also, the method includes monitoring statistics for the read channel, the statistics monitored for the read channel including exponentially averaged mean squared error (MSE) or signal-to-noise ratio (SNR) for each simultaneously read track. Moreover, the method includes performing track-dependent erasure decoding on the encoded data based on detection of one or more time-varying signal quality issues associated with at least one of the plurality of tracks read simultaneously from the magnetic tape medium. The one or more time-varying signal quality issues are determined based on side information about reliability of detected bytes within C1 codewords of each simultaneously-read track.

Abstract: A system, method, and computer-readable medium for performing a selective video manipulation operation for privacy enhancement. More specifically, the selective video manipulation operation detects video transmission protected regions in a scene and applies effects to these regions before transmitting a modified video stream which presents a selectively modified scene. In certain embodiments, the video transmission protected regions include potentially sensitive or distracting regions in the scene. In certain embodiments the selective video manipulation operation automatically detects video transmission protected regions which may be located in at least one of the background of the scene or the foreground attic scene and applies user-configurable effects to the identified regions.

Abstract: A video data processing system and an associated method for generating a summarized video of a recorded video are provided. The method includes the steps of: receiving video data and recording-related information generated during recording video data from at least one source; analyzing the recording-related information and extracting required information from the recording-related information to generate metadata information for the received video data; cropping the received video data to generate cropped video data based on the metadata information; and generating a summarized video of the recorded video data based on the cropped video data.

Abstract: The present invention proposes a method to mark and exploit at least one sequence record of a video presentation played on a multimedia unit, said method comprising the steps of: during the video presentation, receiving a command from a user to mark a currently displayed video sequence, said command initiating the step of: creating a sequence record comprising a time index or frame index, allowing to locate the proper part of the video presentation, and a reference of the video presentation. At a later stage, requesting the edition of the sequence record by: adding textual information which corresponds to the actual sequence, and storing the sequence record.