Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting procession of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: A down sampler 13 down samples a digital signal in the sampling frequency thereof from 96 kHz to 48 kHz on a frame-by-frame basis. The converted signal is compression encoded and output as a main code Im. An up sampler 16 converts a partial signal corresponding to the main code Im to a signal having the original sampling frequency 96 kHz, for example. An error signal between the up sampled signal and an input digital signal is generated. An array converting and encoding unit 18 array converts bits of sample chains of the error signal, thereby outputting an error code Pe. On a decoding side, a high fidelity reproduced signal is obtained based on the main code Im and the error code Pe, or a reproduced signal is obtained based on the main code Im only.

Abstract: An entropy encoder includes an apparatus for producing a data stream which comprises two reference points, of code words of variable lengths, the apparatus comprising a first device for writing at least a part of a code word into the data stream in a first direction of writing, starting from a first reference point, and a second device for writing at least a part of a code word into the data stream in a second direction of writing, which is opposite to the first direction of writing, starting from the other reference point. In particular, when a raster having a plurality of segments is used to write the code words of variable lengths into the data stream, the number of the code words which can be written starting at raster points is doubled, in the best case, such that the data stream of code words of variable lengths is robust toward a propagation of sequence errors.

Abstract: An encryption technique and a decryption technique that are applied to video data that is encrypted on a predetermined domain of compressed video data are disclosed. For example, JPEG2000 compressed video data can be compressed using a wavelet conversion. Compressed video data is input. The input video data is decompressed to a predetermined domain of the compression process, and the decompressed video data is then encrypted. The encrypted video data is then compressed in accordance with the decompression process, and information representing the encrypted compression domain is stored in a predetermined area of the compressed video data.

Abstract: Circuits and methods for providing zero-gap playback of consecutive data streams in portable electronic devices, such as media players, are described. In some embodiments, a circuit includes a decoder circuit configured to receive encoded audio data and to output decoded audio data including data streams associated with a data file and a subsequent data file. Moreover, a predictive circuit, which is electrically coupled to the decoder circuit, is configured to selectively generate additional samples based on samples in the data file, where the additional samples correspond to times after the end of a data stream associated with the data file. Additionally, a filter circuit, which is electrically coupled to the decoder circuit and selectively electrically coupled to the predictive circuit, is configured to selectively combine or blend samples at a beginning of the subsequent data file with the additional samples. Note that the circuit may be included in an integrated circuit.

Abstract: The subject matter includes systems, engines, and methods for generalizing a class of Lempel-Ziv algorithms for lossy compression of multimedia. One implementation of the subject matter compresses audio signals. Because music, especially electronically generated music, has a substantial level of repetitiveness within a single audio clip, the basic Lempel-Ziv compression technique can be generalized to support representing a single window of an audio signal using a linear combination of filtered past windows. Exemplary similarity searches and filtering strategies for finding the past windows are described.

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 invention concerns a method of encoding signals, in particular digitized audio signals, with an encoding device for encoding the signal in an encoding format and a processing device for processing the encoded signal. Methods of that kind are known for example from European patent specification No 290 581. In that case, in the bit rate-reducing encoding of audio signals which are already present in digitized form, for example 48 kHz sampling frequency/16-bit resolution, psycho-acoustic phenomena of the perception of audio signals are used in such a way that the original bit rate of the audio signals is considerably reduced. Such methods are also familiar and standardised under the heading of ‘source encoding’ (ISO 11172 and 11318). The object of the invention is to provide a method of the kind set forth in the opening part of this specification, which resolves the above-indicated problems and in which re-coding operations, once encoding has been effected, are very substantially avoided.

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: A hierarchical lossless encoding and decoding technology for digital signals such as of music data, audio data, or the like. A lossless reproduced signal is made identical to an input signal even when the processing accuracy in an encoding apparatus and the processing accuracy in a decoding apparatus are different from each other. the encoding apparatus transmits low-bit-rate encoded data produced by encoding an input signal, lossless encoded data produced by effecting lossless encoding on a differential signal between a low-bit-rate decoded signal decoded from the low-bit-rate encoded data and the input signal, and corrective information extracted from the low-bit-rate decoded signal to respective input terminals of the decoding apparatus. A low-bit-rate decoder decodes the low-bit-rate encoded data. A lossless decoder decodes the differential signal. A corrector corrects the low-bit-rate decoded signal based on the corrective information.

Abstract: A method of encoding a multichannel signal, such as a stereophonic audio signal, including at least first and second signal components includes transforming at least the first and second signal components by a predetermined transformation into a principal signal including most of the signal energy and at least one residual signal including less energy than the principal signal. The predetermined transformation is parameterized by at least one transformation parameter. The method further includes representing the multichannel signal at least by the principal signal and the transformation parameter.

Abstract: An entropy encoder includes an apparatus for producing a data stream which comprises two reference points, of code words of variable lengths, the apparatus comprising a first device for writing at least a part of a code word into the data stream in a first direction of writing, starting from a first reference point, and a second device for writing at least a part of a code word into the data stream in a second direction of writing, which is opposite to the first direction of writing, starting from the other reference point. In particular, when a raster having a plurality of segments is used to write the code words of variable lengths into the data stream, the number of the code words which can be written starting at raster points is doubled, in the best case, such that the data stream of code words of variable lengths is robust toward a propagation of sequence errors.

Abstract: A digital audio decoder for receiving an encoded audio signal and decoding the audio signal. The digital audio decoder uses both real time computations and calculations pre-stored in a look-up table to decode the encoded audio signal. The digital audio decoder uses fixed point arithmetic but a variable word format to represent the intermediate computations and pre-stored look-up table entries.

Abstract: A device in a subscriber television system receives a stream of frames of information, and each frame of information is carried in at least one network packet that is formatted according to a first protocol. Each network packet carries an application packet that is formatted according to a second protocol, which is a protocol for an application. The device includes application awareness that enables the device to selectively modify the application packets.

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: A header compression/decompression apparatus that improves the throughput of an overall multilayer protocol stack at a network node. In this apparatus, an encoding section 106 compresses multilayer header information included in a protocol data unit on a multilayer protocol stack 101. A session context ID manager 112 generates a session context ID 401 which is formed by integrating information on compression of multilayer header information by the encoding section 106 and, for example, information for identifying a scheme for compressing multilayer header information.

Abstract: Distortion artifacts preceding a signal transient in an audio signal stream processed by a transform-based low-bit-rate audio coding system employing coding blocks are reduced by detecting a transient in the audio signal stream and shifting the temporal relationship of the transient with respect to the coding blocks such that the time duration of the distortion artifacts is reduced. The audio data is time scaled in such a way that the transients are temporally repositioned prior to quantization in a transform-based low-bit-rate audio encoder so as to reduce the amount of pre-noise in the decoded audio signal. Alternatively, or in addition, in a transform-based low-bit-rate audio coding system, a transient in the audio signal stream is detected and a portion of the distortion artifacts are time compressed such that the time duration of the distortion artifacts is reduced.

Abstract: A mobile communication terminal stores a plurality of response messages generated according to respectively different audio compression methods. If the mobile communication terminal is not answered when an incoming call is received, the mobile communication terminal transmits the one of the response messages that has been generated according to the audio compression method designated by a base station.

Abstract: The present invention is a compression method for compressing digital data. The data is strings of digital values, which can be broken down to a series of 1's and 0's. The present inventive method uses a chaotic system to compress the data. The first step in the inventive method is generating a plurality of periodic orbits that correspond to a plurality of control bit strings. Each of the periodic orbits is formed with a series of numeric values. The next step is to convert the numeric values of the periodic orbits to digital data values, similar in form to the data to be compressed. The digital data values of the periodic orbits are then organized to identically match the original digital data values. Then the control bit strings corresponding to the organized digital data values of the periodic orbits are identified and saved in order, such that applying the control bit strings to the chaotic system will regenerate the original data.

Abstract: Windows of the first type and windows of the second type are identified within a frame using energy associated with each short window within the frame. The short windows of the first type and the short windows of the second type are then grouped into two preliminary groups based on the window type of each short window. Further, if the number of short windows in any of the two preliminary groups exceeds a threshold number, the short windows in this large preliminary group are further grouped into at least two more groups.

Abstract: An encoding device (200) includes: a time characteristic extracting unit (203) that specifies a band for a part of a frequency spectrum based on a characteristic of an audio input signal in a time domain; a time transforming unit (204) that transforms a signal in the specified band to a signal according to frequency-time transform; and an encoded data stream generating unit (205) that encodes the signal obtained by the time transforming unit (204) and at least a part of the frequency spectrum, and generates an output encoded data stream from the encoded signal and the encoded frequency spectrum.

Abstract: A method of determining an encoding rate for digital content. According to the method, the a sample of the content data is encoded for a predetermined period of time. The encoding rate is calculated by knowing the size of the sample of content encoded and the length of time necessary for encoding. In another embodiment, the encoding rate calculated for a specific encoding algorithm and encoding bit rate is averaged with any previously stored encoding rate for this encoding algorithm and encoding bit rate. In accordance with another aspect of the invention, an apparatus is described to carry out the above method.

Abstract: A reproducing apparatus according to the present invention includes a thinning-out unit for thinning out part of a plurality of continuous audio digital data, and a conversion unit for simply increasing or decreasing variations in the amplitude of either continuous plural data including the data immediately preceding the thinned data or continuous plural data including the data immediately following the thinned data so that the data immediately preceding the thinned data will be concatenated with the data immediately following the thinned data along a smooth amplitude-varying curve.

Abstract: Digital signal samples X in a floating-point format, each of which is composed of 1 bit of sign, 8 bits of exponent E and 23 bits of mantissa M, are converted through rounding by an integer formatting part 12 into digital signal samples Y in an integer format, the sequence of the digital signal samples Y is losslessly compression-coded by a compressing part 13 into a code sequence Ca, and the code sequence Ca is output. The digital signal samples Y are converted by a floating point formatting part 15 into digital signal samples X? in the floating-point format, a difference signal ?X indicating the difference between the digital signal sample X? and the digital signal sample X is determined by a subtraction part 16, the difference signal ?X is losslessly coded, and the resulting code sequence Cb is output.

Abstract: The total running time of an original multi-channel program signal is altered to generate a time-shortened (or time-lengthened) program signal. The original program signal may be thought of subdividable into a sequence of program signal portions, each program signal portion being further subdividable into a sequence of signal windows. Differencing circuitry determines, for each program siganl portion, a difference value indicative of a best differnce match between an initial signal window in that signal portion and subsequent signal windows in that signal portion in accordance with a predefined criterion. Removal circuitry then deletes from the original multi-channel program signal a multi-window segment of that signal portion, the deleted segment beginning with the initial signal window and ending with the subsequent signal window that generated the best difference match.

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: Media encoding, transmission, and playback processes and structures employ a multi-channel architecture with different audio channels corresponding to different playback rates for a presentation to be transmitted over a network. Audio frames in the various audio channels all correspond to the same amount of time in the original presentation and have frame indexes that identify in the different audio channels the frames corresponding to the same time interval in the presentation. A user can make a real-time change in playback rate causing selection of a channel corresponding to the new playback rate and a frame required for prompt and smooth transition in the playback rate of the presentation. The architecture can additionally provide channels for graphics data such as image data that are displayed according to the index of the audio, and different audio channels with the same playback rate but different compression schemes for use according to available bandwidth on the network.

Abstract: When the quick traverse play back command (or, quick returning play back command) of the audio information by the compression audio information is given to the play back apparatus, by the control of the control section, the special audio according to the temporarily stored data stored in the first storage means is repeatedly played back, instead of the play back of the audio by the ordinary play back operation.

Abstract: The invention provides a method and apparatus for transmitting real-time data in a multi-access system which eliminates clipping of the data while reducing transmission delays. The invention can be incorporated in any multi-access system where transmission resources are allocated when there is information to transmit. In a preferred embodiment, the invention is incorporated in a multi-access wireless system for the upstream transmission of voice from a mobile station to a base station. Each speech segment received at the mobile is shortened by appropriate editing and buffered until transmission. By editing and buffering speech segments as they are received, clipping can be eliminated while reducing transmission delays.

Abstract: Speech at the beginning of a talkspurt in a discontinuous transmission (DTX) packet telephony system is speeded up to help make up for an access delay incurred during channel allocation. Incoming speech frames are buffered, a pitch period for a current portion of the signal is estimated, and then a pitch period=s worth of the signal is cut from that portion. This is continued until the original access delay, as estimated from the time lag between the commencement of voice input for the talkspurt, and notification that a channel is available, is eliminated. The remainder of the talkspurt is then transmitted without such compression.

Abstract: An envelope generator (20), comprises: an input terminal (20a) for having a signal inputted therein; a first integrator (21) for generating intermediate state of envelopes with a first attack time and a first release time in response to changes in level of said signal inputted through said input terminal (20a) to impart said intermediate state of envelopes to said signal; a second integrator (22) for respectively modifying said intermediate state of envelopes into final state of envelopes with a second attack time and a second release time in response to changes in level of said signal imparted said intermediate state of envelopes; and an output terminal (20d) for outputting said signal with said final state of envelopes therethrough. The envelope generator (20) thus constructed can make gain signal follow rapid fluctuations in level of an audio signal, and can impart a relatively high quality for compressing and expanding level of the audio signal not to break in shape.

Abstract: An MPEG Optimization Software (MOPSW) for maximizing a Video Compression Ratio (VDCR) while maintaining a good output Video Image Quality (VDIQ) of an Input MPEG file (IMPEG) is disclosed. The IMPEG has a set of adjustable MPEG Control Parameters affecting both its VDCR and VDIQ. In a specific embodiment with application to MPEG2 files, the MOPSW employs the following optimized set of adjustable MPEG Control Parameters: Set Video Size=(x)×(Minutes of Video) where x=12 through 18; Maximum BITRATE=2200 through 3300; Maximum Average BITRATE=2200; and Minimum BITRATE=300 and achieved a VDCR that is at least 200% higher than what is typically available from current DVD suppliers in the art while maintaining a good output VDIQ for a representative set of movie titles. An associated method of iteratively determining the optimized set of adjustable MPEG Control Parameters is also presented.

Abstract: An audio signal processing method wherein it is detected whether the data supplied from an optical disk reproduction apparatus or the like has continuous zero data for a predetermined period of time, and in the case where zero data continue for the predetermined period of time, it is determined that compressed audio data is involved, and the supplied data is decoded.

Abstract: A sound signal encoding apparatus for encoding two different sound signals, comprising: compression level calculating means for calculating a compression level for each of the sound signal sections; compression level judging means for judging whether or not the calculated compression level for each of the sound signal sections exceeds a predetermined threshold compression value; frequency components encoding means for encoding the quantized frequency components for each of the sound signal sections to a multiplexed bit stream with a predetermined bit rate under two different states consisting of the first state in which the frequency components for each of the sound signal sections are compressed by the first frequency components compressing means when the compression level judging means is operative to judge that the compression level for each of the sound signal sections exceeds the predetermined threshold compression value and the second state in which the frequency components for each of the sound signal

Abstract: A method for summarizing a video first detects audio peaks in a sub-sampled audio signal of the video. Then, motion activity in the video is extracted and filtered. The filtered motion activity is quantized to a continuous stream of digital pulses, one pulse for each frame. If the motion activity is greater than a predetermined threshold the pulse is one, otherwise the pulse is zero. Each quantized pulse is tested with respect to the timing of rising and falling edges. If the pulse meets the condition of the test, then the pulse is selected as a candidate pulse related to an interesting event in the video, otherwise the pulse is discarded. The candidate pulses are correlated, time-wise to the audio peaks, and patterns between the pulses and peaks are examined. The correlation patterns segment the video into uninteresting and interesting portions, which can then be summarized.

Abstract: A method and apparatus for determining masked thresholds for a perceptual auditory model used, for example, in a perceptual audio coder, which makes use of a filter bank structure comprising a plurality of filter bank stages which are connected in series, wherein each filter bank stage comprises a plurality of low-pass filters connected in series and a corresponding plurality of high-pass filters applied to the outputs of each of the low-pass filters, and wherein downsampling is advantageously applied between each successive pair of filter bank stages. In accordance with one illustrative embodiment, the filter bank comprises low order IIR filters. The cascade structure advantageously supports sampling rate reduction due to the continuously decreasing cutoff frequency in the cascade. The filter bank coefficients may advantageously be optimized for modeling of masked threshold patterns of narrow-band maskers, and the generated thresholds may be advantageously applied in a perceptual audio coder.

Abstract: An audio playback/recording apparatus includes an audio input processing section which receives analog audio data, and converts the analog audio data to digital audio data; a playback/recording processing section which compresses digital audio data output from the audio input processing section and stores the compressed digital audio data into a RAM and which decompresses the compressed digital audio data according to attribution data indicating a type of compression; an audio output processing section which receives the decompressed digital audio data, converts the decompressed digital audio data to analog audio data, and outputs the analog audio data to an output apparatus; and an external recording circuit section which records compressed digital audio data stored in the RAM into an external recording medium, reads out the compressed digital audio data, and stores the data into the RAM.

Abstract: According to the present invention, a voice transceiver is provided which is characterized in comprising: an input mechanism for inputting compressed voice codes of analog data; an expansion unit for digitalizing the compressed voice codes, and expanding and outputting digital voice data; a buffer for storing the digital voice data; a detection unit for detecting the quantity of data of the digital voice data stored in the buffer, and outputting a detection signal as the detection result; a converter for converting the digital voice data into analog voice data based on a detection signal; and a speaker for emitting the analog voice data into the air.

Abstract: An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a pre-determined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

Abstract: A multiplexed audio data decoder apparatus is provided in which integration of an audio decoder is easy, and has a high flexibility when the number of the formats to be processed is increased or when the specification is changed. In an external ROM 60 there are accumulated a plurality of decoding program codes corresponding to respective plural methods for compressing and encoding. A controller means 50 transfers the decoding program code corresponding to the method for compressing and encoding after changing thereof, from the external ROM 60 to an internal RAM 25. A DSP 22 starts decoding processing by using the decoding program code which is transmitted into the internal RAM 25.

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 speech encoding or decoding arrangement (711, 721, 811, 821) comprises a speech signal input and a multiple mode speech encoder (402) or decoder (411) for encoding or decoding speech signals coupled to the speech signal input selectabily with a first encoding or decoding mode associated with a first bandwidth or a second encoding or decoding mode associated with a second bandwidth. It comprises a soft bandwidth switching block (401, 412, 500) with an input (IN) and an output (OUT). In an encoding arrangement the input (IN) is coupled to the speech signal input and the output (OUT) is coupled to the multiple mode speech encoder (402). In a decoding arrangement the input (IN) is coupled to the multiple mode speech decoder (411) and the output (OUT) is the output of the decoding arrangement.

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: Encoding an acoustic source signal such that a signal {circumflex over (z)} reconstructed from the encoded information has a perceptually high sound quality. The acoustic source signal is encoded into at least one basic coded signal that represents perceptually significant characteristics of the acoustic signal. The encoder can include at least one spectral smoothing unit which receives at least one of the signal components on which the basic coded signal is based and generates in response thereto a corresponding smoothed signal component. At least one enhanced coded signal is then produced from the corresponding smoothed signal. component for transmission. A receiver receives at least one estimate {circumflex over (P)}E of the transmitted signal(s), and a spectral smoothing unit in the receiver produces, on basis of a primary spectrum Ŷ decoded from the at least one received estimate {circumflex over (P)}E, a smoothed primary decoded spectrum ŶE.

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 and apparatus for determining parameters for coupling of channels in a digital audio encoder. A frequency range of two audio channels is coupled together in a coupling channel, and a systematic method of determining optimum coupling parameters is employed. Sub-bands of the channels are processed individually, and a measure of the power of each sub-band is used to determine a coupling coefficient generation scheme for each individual sub-band. Adjacent ones of the sub-bands using the same coupling scheme are combined to form bands in the coupling channel, which dictate the generation of the coupling coordinates for the audio channels. The arrangement of the sub-bands in bands also facilitates the generation of phase flags for each band, on the basis of the coupling scheme used in the band.

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: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.