Abstract: An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding to the pilot reference value and obtaining the data using the pilot reference value and the pilot difference value.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.

Abstract: A method of decoding audio data, encoded in multiple DIF blocks in a Digital Video (DV) data stream, and outputting said audio data as a PCM frame, includes fetching a single Digital Interface Frame (DIF) block from the DV data stream. A first byte in the single DIF block is de-shuffled to determine its index (n) in the PCM frame. Each byte in the in the single DIF block is de-shuffled to determine its respective index (n) in the PCM frame. The de-shuffled data is written into the PCM frame for output if the present DIF block is the last in the present DV frame. Subsequent DIF blocks in the DV frame are processed in the manner described above.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: Aspects of a method and system for multi-band amplitude estimation and gain control in an audio CODEC are provided. In this regard, an audio signal may be filtered and delayed to generate one or more sub-band signals, a gain may be applied to each sub-band signal to generate one or more level adjusted sub-band signals, and the one or more level adjusted signals may be added to a delayed version of the audio signal. The gain applied to a particular one of the one or more sub-band signals may be controlled based on a detected amplitude of a summed signal derived by summing the particular one of the one or more sub-band signals and a corresponding one of the one or more level-adjusted sub-band signals.

Abstract: A multi-channel synthesizer for generating at least three output channels using an input signal having at least one base channel, the base channel being derived from the original multi-channel signal, the input signal further including at least two different up-mixing parameters, and an up-mixer mode indication indicating, in a first state that a first up-mixing rule is to be performed, and, indicating, in a second state, that a different second up-mixing rule is to be performed, uses an up-mixer for up-mixing the at least one base channel using the at least two different up-mixing parameters based on the first or the second up-mixing rule in response to the up-mixer mode indication so that the at least three output channels are obtained.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

Abstract: In a method for generating a scalable data stream from one or several blocks of output data of a first encoder and from one or several blocks of output data of a second encoder a determining data block for a current section of an input signal is written. In addition, output data of the second encoder representing a preceding section of the input signal are written in transmission direction from an encoder to a decoder after the determining data block. When the output data of the second encoder are written for a preceding section of the input signal, the output data of the second encoder are written representing the current section of the input signal. In order to signalize where the output data of the second encoder for the preceding section end and where the output data of the second encoder for the current section begin, buffer information is written into the scalable data stream.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism (703a) on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded (703b) and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: A control logic means preferably for a receiver comprising a jitter buffer means adapted to receive and buffer incoming frames or packets and to extract data frames from the received packets, a decoder connected to the jitter buffer means adapted to decode the extracted data frames, and a time scaling means connected to the decoder adapted to play out decoded speech frames adaptively. The control logic means comprises knowledge of whether a state recovery function is available and is adapted to retrieve at least one parameter from at least one of the jitter buffer means, the time scaling means, and the decoder, to adaptively control at least one of an initial buffering time of said jitter buffer means, the knowledge of the availability of the state recovery function, and a time scaling amount of said time scaling means from the time scaling means or the decoder.

Abstract: In a method of producing a scalable data stream of at least two blocks of output data of a first coder and a block of output data of a second coder, wherein the at least two blocks of output data of the first coder together represent a current section of an input signal in the first coder, and wherein the block of output data of the second coder represents the same current section of the input signal, a determination data block for the current section of the input signal is written. In addition, the block of output data of the second coder, in the direction of transfer from a coding device to a decoding device, is written after the determination data block for the current section of the input signal.

Abstract: An audio processing method utilized to generate an audio stream. An audio frame includes N frequency subbands. An Ith frequency subband among the N frequency subbands includes M audio samples and has an Ith psychoacoustic masking value. First, an Ith offset of the Ith frequency subband is calculated. Then, the Ith psychoacoustic masking value and the Ith offset are inputted into a projection formula to generate an Ith projection value. According to the Ith projection value and a limit range, an Ith scale factor is determined. Subsequently, the M audio samples in the Ith frequency subband are adjusted according to the Ith scale factor.

Abstract: A method and an apparatus analogically output low-resolution digital signals to achieve an equal analog output result of a high-resolution digital signal under a request of signal quality. The low-resolution digital signals are compensatively output multiple times to gain the energy thereof equal to the energy output by the high-resolution digital signal. Therefore, a digital-to-analog conversion with fewer bits satisfies a higher demand for accuracy generally achieved by a digital-to-analog conversion with more bits.

Abstract: The invention provides a method for obtaining related information about a media program containing an audio signal, comprising: an embedding step of embedding an audio watermark containing identification information of the media program into the audio signal of the media program; a transmitting step of transmitting the media program; a recording step of recording by a user a portion of the audio signal of the media program embedded with the audio watermark; an extracting step of extracting the audio watermark from the recorded portion of the audio signal to obtain the identification information and providing the identification information to a server storing the related information; and a related information providing step of providing the user with the related information about the media program according to the identification information from the server, wherein the embedding step comprises: identification information pre-processing step of performing a pre-processing on the identification information of th

Abstract: In a telecommunications system, an arithmetic logic unit (ALU) that receives an input signal. The input signal includes a digital signal representative of an analog signal. The ALU selectively performs compression and decompression on the digital signal. The ALU comprises the following elements. A standard ALU component performs standard ALU operations on the input signal. An encoding unit selectively performs compression on the digital signal. A decoding unit selectively performs decompression on the digital signal. An instruction decoder receives and decodes an ALU instruction. An output selector selects a result from one of the standard ALU component, the encoding unit, and the decoding unit in accordance with the decoded instruction and provides the result as an output.

Abstract: An audio processing unit that provides a high-sound-quality reproduction even when a band property of an input encoded signal drops at or below a Nyquist frequency. The audio processing unit has a band-variable bandpass filter 1500; a band determining section 1700 for determining a passband of the band-variable bandpass filter with respect to a band extension component from an extended band generating section 1300 by using decoding information from a decode-processing section 1100 as a band determination information; and a bandpass filter controller 1600 for controlling the passband of the band-variable bandpass filter in accordance with an indication from the band determining section.

Abstract: A system for transmitting audio signals over a telecommunications link generates the signals as two or more alternative feeds, for example at different data rates. The two feeds are encoded using coding methods having a frame structure with different frame lengths. To facilitate switching between the two, the input signal is notionally divided into temporal portions and each is coded by taking it, plus enough of the next (or preceding) portion to make up a whole number of frames, and encoding it, whereby the encoded portions overlap—at least for one of the feeds. The overlap is lost upon decoding by discarding duplicate material.

Abstract: A compressed-code generating method that is used for compressing information on characters including numerical data, sound and images, and a compressed-code expanding method that is used for restoring and expanding the compressed code generated by using the compressed-code generating method to the original information. Bit strings {y}1 and {y}2 are obtained respectively from a bit string {y} of information to be compressed. A reversible loop that exists in chaos is operated to these obtained bit strings, thereby to execute a reversible compression/expansion of the information using the chaos.

Abstract: In a voice communication system 1, a gateway server 4 receives IP packets from the Internet, converts PCM voice data in the IP packets into AMR encoded voice data frames, and transmits to a mobile terminal 7. During the propagation to the gateway server 4, there is a possibility of loss of IP packets and crucial bit error in IP packets. In that case, the gateway server 4 puts “No data” data on frames as voice encoded data for the IP packets in question and sends it to the mobile terminal 7. The “No data” data is a target of concealment.

Abstract: An audio decoding device is provided for decoding NA (where NA>1) channels of audio signals by a sub-band synthesis operation using sub-band synthesis filter data and sub-band signal data. The decoding device includes a first memory section for storing MA (where MA<NA) channels of the sub-band synthesis filter data and the sub-band signal data, a second memory section for storing at least some of NA channels, an operation section for receiving encoded audio data and decoding the encoded audio data into sub-band signal data, and a data transfer section for, switching, by MA channels, the sub-band synthesis filter data and the sub-band signal data in the first memory section and the second memory section.

Abstract: A method of normalizing received digital audio data includes decomposing the digital audio data into a plurality of sub-bands and applying a psycho-acoustic model to the digital audio data to generate a plurality of masking thresholds. The method further includes generating a plurality of transformation adjustment parameters based on the masking thresholds and desired transformation parameters and applying the transformation adjustment parameters to the sub-bands to generate transformed sub-bands.

Abstract: An improved representation of transients in audio signals comprises modifying transient locations in such a way that a transient can occur only at a beginning of a sinusoidal segment. The modification procedure comprises the steps: detecting a beginning and an end of a transient using an energy-based approach with two sliding rectangular windows; moving samples between the beginning and the end of the transient to the locations specified by the segmentation used; and time-warping the signal parts in between the transients in order to fill the intervals between the modified transients.

Abstract: A digital audio signal to be replayed is processed in a waveform thereof. A frequency bandwidth of the audio signal is expanded through conversion of a sampling frequency, and then the audio signal is low-pass-filtered with a low-pass cut-off frequency corresponding to the converted sampling frequency. An interval of time between two waveform peaks of the audio signal is detected, and then difference data between current data of the audio signal and past data thereof is calculated. The difference data are subject to weighting depending on the interval, and then output data are produced based on both the low-pass-filtered audio signal and the weighted difference data. This processing, which can be realized by activation of software, improves audio quality when compressed audio data is replayed.

Abstract: An interleaved delay line for use in phase locked and delay locked loops is comprised of a first portion providing a variable amount of delay substantially independently of process, temperature and voltage (PVT) variations while a second portion, in series with the first portion, provides a variable amount of delay that substantially tracks changes in process, temperature, and voltage variations. By combining, or interleaving, the two types of delay, single and dual locked loops constructed using the present invention achieve a desired jitter performance under PVT variations, dynamically track the delay variations of one coarse tap without a large number of delay taps, and provide for quick and tight locking. Methods of operating delay lines and locked loops are also disclosed.

Abstract: A method and a device for defining bit allocation table in processing audio signals are provided. The provided method and device can save storage bits and provide light quality as well. In the first step, the total number of bits for storing audio signals is determined. Then the psychoacoustic model provides many signal-to-mask ratios according to the audio signals. At last, the quantizer quantizes the signal-to-mask ratios to generate several quantized levels each of which corresponds to a bit allocation value to define the table of bit allocation. Therefore, fewer or no storage bits are provided for unimportant subbands and signal frames, that is, the efficiency and quality of transmission of audio signals can be raised.

Abstract: Voiceband compression techniques are employed in order to enable an RF telecommunications base station to accommodate data signals of high speed voiceband modems and FAX machines. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s and operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Time slot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A media processing apparatus includes an input/output processing unit that performs input/output processing that is asynchronously caused by external factors and a decode processing unit that operates in parallel with the input/output processing unit and mainly performs decode processing for a data stream stored in a memory. The input/output processing receives an asynchronous input of a data stream from outside, stores the data stream into the memory, and supplies a data stream stored in the memory to the decode processing unit. The decode processing unit includes a sequential processing unit and a routine processing unit. The sequential processing unit performs header analysis for compressed video data in the data stream and a complete decode for compressed audio data in the data stream. Based on the results of the header analysis, the routine processing unit performs all of the decoding of the compressed video data aside from the header analysis.

Abstract: A method of compressing an audio signal can include accepting input samples of the audio signal wherein the input samples include non-zero input samples. A logarithm of each of the non-zero input samples of the audio signal can be calculated. Compressed output samples for each non-zero input sample can then be determined based on the logarithm of each respective non-zero input sample. Preferably, a linear relationship may exist between logarithms of the non-zero input samples and logarithms of the corresponding compressed output samples. A logarithm of each compressed output sample, corresponding to a non-zero input sample, may be based on a product of a logarithm of each corresponding non-zero input sample and a compression factor. Related devices and computer program products are also discussed.

Abstract: A new approach to data embedding within ITU G.722 and ITU G.711 based upon the method of types and universal classification is disclosed. A secondary data sequence is embedded in the original (host) data stream using the method of types. The embedded data is extracted using a type-based universal receiver, with or without the use of a key. The choice of type and rate for the embedded data is based upon an analysis of portions of the original ITU G.722 or ITU G.711 coded data stream. The universal receiver learns the type from the received data alone, and hence, there is no side information required as in previous data embedding techniques. The embedding process and the receiver may both be data adaptive, so the original data stream can be reconstructed at the decoder without error.

Abstract: A fast code length search method for determining the length of a code in a codebook, wherein the method is especially suited for MPEG-compliant audio encoding. A code length table is created which stores pre-calculated code lengths, including any sign bits and linear extension bits necessary, for data value pairs or quadruples. In one embodiment, two code length tables are created, one for determining the code lengths of the codes used for the ones region, and a second code length table for the big values region. When a code length determination is made, the value is simply read from the table, instead of being calculated each time.

Abstract: A method for transferring audio data and audio-related information includes generating second audio data from first audio data, transmitting second audio data and audio-related information associated with the second audio data, and receiving the second audio data and audio-related information which includes information on a sampling frequency of the first audio data.

Abstract: For a left-channel input audio signal, digital filters 30LL and 30LR are provided for reproduction of impulse responses used for locating an acoustic image outside the head of a listener, and for a right-channel input audio signal, digital filters 30RL and 30RR are provided for reproduction of impulse responses for locating an acoustic image outside the listener's head. An addition circuit 7L adds output signals of the digital filters 30LL and 30RL to each other and a second addition circuit 7R adds output signals of the digital filters 30LR and 30RR to each other. Output signals of the addition circuits 7L and 7R are supplied to a headphone, whereby the reproduction acoustic images of the left-channel and right-channel input audio signals are located outside the listener's head. The sampling rate in each of the digital filters 30LL-30RR is changed depending on the section of the response time of the associated impulse response to be reproduced.

Abstract: Two related voiceband compression techniques are employed in order to enable an RF telecommunications system to accommodate data signals of high speed voiceband modems and FAX machines. A High Speed Codec enables the telecommunications system to pass voiceband modem and FAX transmissions at up to 9.6 kb/s. An Ultra-High Speed Codec supports voiceband modem and FAX transmissions up to 14.4 kb/s. The High Speed Codec operates using three 16-phase RF slots or four 8-phase RF slots, and the Ultra-High Speed Codec operates using four 16-phase RF slots. Because these codecs transmit information over several RF slots which can be contiguous, the slots within RF communication channels are dynamically allocated. The Dynamic Timeslot/Bandwidth Allocation feature detects and monitors the data transmission and forms a data channel from the necessary number of slots.

Abstract: A method and a system is provided for embedding and detecting additional information, such as copyright information, in audio data, so that a modification in the sonic quality due to the embedding is imperceptible to human beings, and does not drastically deteriorate the sonic quality.

Abstract: A sound signal processing apparatus including extracting means for extracting from a composite sound signal, representing multiple sound sequences, digital sound signals corresponding to a portion of the composite sound signal. Each of the digital sound signals is individually sampled. Also included is a signal converter for converting the digital sound signals which have been extracted into analog sound signals.

Abstract: In a coded speech decoding system, an n-channel time domain speech signal is converted to a frequency domain speech signal. A predetermined weighting adding process is executed on the frequency domain speech signal for each of a plurality of different transfer functions. The frequency domain speech signal obtained through the weighting adding process is converted to an m-channel (m<n) time domain speech signal. A predetermined windowing processing is executed on the time domain speech signal.

Abstract: An audio information bit stream including audio control bits and audio data bits is processed for transmission in a communication system. The audio data bits are first separated into n classes based on error sensitivity, that is, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. Each of the n different classes of audio data bits is then provided with a corresponding one of n different levels of error protection, where n is greater than or equal to two. The invention thereby matches error protection for the audio data bits to source and channel error sensitivity. The audio control bits may be transmitted independently of the audio data bits, using an additional level of error protection higher than that used for any of the n classes of the audio data bits. Alternatively, the control bits may be combined with one of the n classes of audio data bits and provided with the highest of the n levels of error protection.

Abstract: A data processing device uses a portion of a random access memory as an output buffer for holding a frame of PCM sample data which is being output after being processed by a processing unit within the processing device. Fine grained synchronization between a reference clock and a stream of PCM data frames is provided by transferring only a portion of selected frame of PCM sample data PCM(n+1), in response to a time difference 971. A breakpoint address is determined to delineate the portion of the selected frame that is to be transferred. A sorted list of the addresses of the discontinuities is maintained in breakpoint queue. Since the buffer is managed in a FIFO manner, a single breakpoint register is sufficient to monitor addresses as they are provided by an address register for accessing the random access memory. When a breakpoint is detected, the breakpoint queue and the breakpoint register is updated by an update task 802.

Abstract: A portable audio player stores a large amount of compressed audio data on an internal disk drive, and loads a portion of this into an internal random access memory (RAM) which requires less power and less time to access. The audio player plays the data stored in RAM and monitors the amount of unplayed data. When the amount of unplayed data falls below a threshold, additional data is copied from the disk drive into RAM. Because the time necessary to copy a block of data from the disk drive to RAM is much less than the amount of time it takes to play the same block of audio data from RAM, this approach minimizes the amount of time that the disk drive must be operated, and thus minimizes the amount of power consumed by the system.

Abstract: A multi-channel audio compression technology is presented that extends the range of sampling frequencies compared to existing technologies and/or lowers the noise floor while remaining compatible with those earlier generation technologies. The high-sampling frequency multi-channel audio is decomposed into core audio up to the existing sampling frequencies and a difference signal up to the sampling frequencies of the next generation technologies. The core audio is encoded using a first generation technology such as DTS, Dolby AC-3 or MPEG I or II such that the encoded core bit stream is fully compatible with a comparable decoder in the market. The difference signal is encoded using technologies that extend the sampling frequency and/or improve the quality of the core audio. The compressed difference signal is attached as an extension to the core bit stream. The extension data will be ignored by the first generation decoders but can be decoded by the second generation decoders.

Abstract: An adaptive linear predictor is used to predict samples, and residuals from such predictions are encoded using Golomb-Rice encoding. Linear prediction of samples of a signal which represents digitized sound tends to produce relatively low residuals and those residuals tend to be distributed exponentially. Accordingly, linear prediction combined with Golomb-Rice encoding produces particularly good compression rates with very efficient and simple implementation. The accuracy of the linear predictor is improved by including, in the prediction of a current sample of a first channel of the digitized signal, look-ahead sample data from a corresponding second channel of the digitized signal. For example, prediction of a right channel sample of a digitized, stereo, audio signal is improved by inclusion of look-ahead left channel sample data in the right channel sample predictor.

Abstract: An audio decoder is described which supports simple sound-effect generation. The audio decoder includes a direct access pulse code modulation (PCM) first-in-first-out buffer (FIFO) to support simple sound effect generation. In one embodiment, the audio decoder additionally includes an input buffer, a decoding module, and an output interface. The input buffer buffers incoming data frames for the decoding module to retrieve and convert to a sequence of decoded audio samples. The FIFO is configured to receive and buffer audio sound effect samples from a control component external to the audio decoder. The output interface is configurable to retrieve decoded audio samples from the decoding module and audio sound effect samples from the FIFO. Any retrieved audio sound effect samples are included in a digital audio output signal provided by the output interface. The digital audio output signal may be provided directly to a digital-to-analog converter for sound reproduction.

Abstract: An audio signal is recorded in a semiconductor memory in a plurality of hierarchical levels, with the lowest level being adequate for reproduction with a certain reduced degree of fidelity. Successively higher hierarchial levels provide successively greater fidelity when reproduced. When the memory has been determined to have reached maximum capacity, recording continues by overwriting the highest hierarchical level of data currently stored in the memory with lower hierarchical levels of new data. A code is recorded in the memory, indicating the number of hierarchical levels recorded therein, for subsequent reproduction. The audio signal can furthermore be recorded in variable-length frames and reproduced at high speed by reading every N-th frame, N being a positive integer, or by reading only frames having at least a certain minimum length.