Abstract: A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.

Abstract: Memory resources can be optimized by dynamically determining a threshold value of a storage device used for buffering in accordance with a compression rate of data for streaming reproduction. A data reproduction device for temporarily storing compressed data that is downloaded from a server and sequentially performing the streaming reproduction, wherein the amount of data stored in a HDD is optimized by changing and setting the threshold value in accordance with the compression rate of the compressed data.

Abstract: The invention is aimed at improving the quality of the filtering by transfer functions of HRTF type of signals (L, R) compressed in a transformed domain, for binaural playing on two channels (L-BIN, R-BIN), using a combination of HRTF filters (hL,L, hL,R) including a decorrelated version (HRTF-C*, HRTF-E*) of a few of these filters. For this purpose, a decorrelation cue is given with spatialization parameters (SPAT) accompanying the compressed signals (L, R). The invention makes it possible to improve the broadening in the binaural rendition of audio scenes initially in a multi-channel format.

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: A lossless audio encoding/decoding method, medium, and apparatus. The lossless audio encoding method includes converting an audio signal in a time domain into an audio spectral signal with an integer in a frequency domain, mapping the audio spectral signal in the frequency domain to a bit plane signal according to its frequency, and losslessly encoding binary samples of bit planes using a probability model determined according to a predetermined context. The lossless audio decoding method includes extracting a predetermined lossy bitstream and an error bitstream from error data by demultiplexing an audio bitstream, the error data corresponding to a difference between lossy encoded audio data and an audio spectral signal with an integer in a frequency domain, lossy decoding the extracted encoded lossy bitstream, losslessly decoding the extracted error bitstream, and restoring the original audio frequency spectral signal using the decoded lossy bitstream and error bitstream.

Abstract: An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors.

Abstract: To encode multi-channel digital data by adjusting the number of bits allocated to each channel to perform entropy coding of the multi-channel data, there is provided a multi-channel encoder including n encoders for audio data from n channels and an inter-channel bit allocator that allocates the number of bits usable for each channel on the basis of the provisional number of in-use bits from each of the encoders. Each of the encoders performs entropy coding on the basis of the provisional number of quantizing steps, outputs the provisional number of in-use bits resulting from summing of a code length of each unit of coding, and adjusts the number of in-use bits by updating the quantizing steps correspondingly to the number of bits supplied based on the provisional number of in-use bits.

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: Systems and processes for transmission of multi-channel audio from a sender to one or more recipients. Multi-channel audio is encoded with a plurality of different dialog channels. The encoded multi-channel audio and dialog channels can be compressed to facilitate transmission with limited bandwidth. A recipient can select a desired dialog channel from the plurality of dialog channels transmitted. A receiver side decoder can reconstruct a multi-channel audio with the selected dialog for playback. The plurality of dialog options can include different languages, different dialects, different accents, and/or different viewpoints/biases.

Abstract: Disclosed is a system and method for channel decoding speech frames in a receiver capable of multiple (M) codec modes, wherein channel encoded speech frames include an inband bit portion and a speech portion. An inband bit decoder decodes the inband bit portion (700) of a received frame to obtain confidence levels associated with each of the M codec modes. Using these confidence levels, the codec modes are ordered from most to least likely. The speech frame is then decoded by a channel decoder using the most likely codec mode (704). A frame determination check (720) is performed to determine the quality of the decoded speech frame. If the decoded speech frame is determined to be of poor quality, then the channel decoding process is repeated using the next most likely codec mode (736) corresponding to the next highest inband bit decoding confidence level. This process is repeated until a good speech frame is decoded or some exit criteria is reached.

Abstract: There is described a multi-channel encoder (10; 600) for processing input signals conveyed in N input channels to generate corresponding output signals conveyed in M output channels together with complementary parametric data; M and N are integers wherein N>M. The encoder (10; 600) includes a down-mixer for down-mixing the input signals to generate the corresponding output signals, the encoder also comprising an analyser for processing the input signals to generate the parameter data, said parametric data describing mutual differences between the N channels of input signal to allow for regenerating during decoding one or more of the N channels of input signals from the M channels of output signal. Such an encoder (10; 600) is capable of providing highly efficient data encoding and also of being backwards compatibility with relatively simpler decoders having fewer than N decoding output channels. The invention also concerns decoders (800) compatible with such a multi-channel encoder (10; 600).

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: Techniques and tools for selectively using multiple entropy models in adaptive coding and decoding are described herein. For example, for multiple symbols, an audio encoder selects an entropy model from a first model set that includes multiple entropy models. Each of the multiple entropy models includes a model switch point for switching to a second model set that includes one or more entropy models. The encoder processes the multiple symbols using the selected entropy model and outputs results. Techniques and tools for generating entropy models are also described.

Abstract: The invention relates to a method for supporting an encoding of an audio signal, wherein a first coder mode and a second coder mode are available for encoding a respective section of an audio signal. The second coder mode enables a coding of a respective section based on a first coding model, which requires for an encoding of a respective section only information from the section itself, and based on a second coding model, which requires for an encoding of a respective section in addition an overlap signal with information from a preceding section. After a switch from the first coder mode to the second coder mode, always the first coding model is used for encoding a first section of the audio signal. This section can then be employed to generate an artificial overlap signal for a subsequent section, which is possibly to be encoded with the second coding model.

Abstract: This invention relates to a method, a device and a software application product for N-level quantization of vectors, wherein N is selectable prior to said quantization from a set of at least two pre-defined values that are smaller than or equal to a pre-defined maximum number of levels M. A reproduction vector for each vector is selected from an N-level codebook of N reproduction vectors that are, for each N in said set of at least two pre-defined values, represented by the first N reproduction vectors of the same joint codebook of M reproduction vectors. The invention further relates to a method, a device and a software application product for retrieving reproduction vectors for vectors that have been N-level quantized, to a system for transferring representations of vectors, to a method, a device and a software application product for determining a joint codebook, and to such a joint codebook itself.

Abstract: Several encoders at a broadcast system encode the same audio content. Packets from the resulting streams are immediately decoded and compared against the packets of the original audio stream. The broadcast system dynamically selects the codec that performs the best for the audio in any given packet. The packet produced by the encoder of the best-performing codec devices is selected to be broadcasted/transmitted.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: There is provided an audio encoding device capable of maintaining continuity of spectrum energy and preventing degradation of audio quality even when a spectrum of a low range of an audio signal is copied at a high range a plurality of times. The audio encoding device (100) includes: an LPC quantization unit (102) for quantizing an LPC coefficient; an LPC decoding unit (103) for decoding the quantized LPC coefficient; an inverse filter unit (104) for flattening the spectrum of the input audio signal by the inverse filter configured by using the decoding LPC coefficient; a frequency region conversion unit (105) for frequency-analyzing the flattened spectrum; a first layer encoding unit (106) for encoding the low range of the flattened spectrum to generate first layer encoded data; a first layer decoding unit (107) for decoding the first layer encoded data to generate a first layer decoded spectrum, and a second layer encoding unit (108) for encoding.

Abstract: Techniques and tools for representing, coding, and decoding scale factor information are described herein. For example, during encoding of scale factors, an encoder uses one or more of flexible scale factor resolution selection, spatial prediction of scale factors, flexible prediction of scale factors, smoothing of noisy scale factor amplitudes, reordering of scale factor prediction residuals, and prediction of scale factor prediction residuals. Or, during decoding, a decoder uses one or more of flexible scale factor resolution selection, spatial prediction of scale factors, flexible prediction of scale factors, reordering of scale factor prediction residuals, and prediction of scale factor prediction residuals.

Abstract: A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and/or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and/or second up-mix signals S1 and S2 from the downmix signal d according to a computation representable by ( S 1 S 2 ) = D - 1 ? { ( 1 C ) ? d + H } , with “1” denoting—depending on the number of channels of d—a scalar, or an identity matrix, and D?1 being a matrix uniquely determined by a downmix prescription according

Abstract: An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.

Abstract: An BTSC encoder with an improved digital filter structure substantially implemented on a single CMOS integrated circuit is described. By cascading first and second order allpass filter structures to form a higher order digital filter, such as a Cauer low pass filter, limit cycle oscillations are reduced or eliminated, word-length growth from one stage to the next is contained, and a more efficient overall filter structure and performance is obtained.

Abstract: A digital audio encoding method using an advanced psychoacoustic model is provided. The audio encoding method including determining the type of a window according to the characteristic of an input audio signal; generating a complex modified discrete cosine transform (CMDCT) spectrum from the input audio signal according to the determined window type; generating a fast Fourier transform (FFT) spectrum from the input audio signal, by using the determined window type; and performing a psychoacoustic model analysis by using the generated CMDCT spectrum and FFT spectrum.

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: Analysis and synthesis filter banks such as those used in audio and video coding systems are each implemented by a hybrid transform that comprises a primary transform in cascade with one or more secondary transforms. The primary transforms for the filter banks implement an analysis/synthesis system in which time-domain aliasing artifacts are cancelled. The secondary transforms, which are in cascade with the primary transforms, are applied to blocks of transform coefficients. The length of the blocks is varied to adapt the time resolution of the analysis and synthesis filter banks.

Abstract: An integer transform, which provides integer output values, carries out the TDAC function of a MDCT in the time domain before the forward transform. In overlapping windows, this results in a Givens rotation which may be represented by lifting matrices, wherein time-discrete sampled values of an audio signal may at first be summed up on a pair-wise basis to build a vector so as to be sequentially provided with a lifting matrix. After each multiplication of a vector by a lifting matrix, a rounding step is carried out such that, on the output-side, only integers will result. By transforming the windowed integer sampled value with an integer transform, a spectral representation with integer spectral values may be obtained. The inverse mapping with an inverse rotation matrix and corresponding inverse lifting matrices results in an exact reconstruction.

Abstract: A synthesis filter for an MPEG audio decoder, and a decoding method thereof. The synthesis filter for an MPEG audio decoder includes a butterfly computation part which performs a butterfly computation with respect to an input MPEG audio subband sample by a predetermined unit, a matrix computation part which performs a matrix computation using a result of the butterfly computation, a first buffer which stores a result of the matrix computation according to a predetermined address, a window computation address generation part which generates first and second addresses with respect to the first buffer to use for each cycle of the window computation, a window computation part which outputs the matrix computation result stored to the first and second addresses to perform the window computation, and an output part which outputs a result of the window computation.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: The present invention intends to render quantization noise virtually imperceptible for a user and to prevent reduction in frequency resolution and reduction in encoding efficiency.

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: An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L?R signals.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information and partitioning information from the configuration information. The coding type information indicates an entropy coding scheme used in encoding the audio signal, and the partitioning information indicates a sub-block partition scheme by which the block is divided into sub-blocks. Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the sub-block partitioning scheme. The number of the sub-blocks is determined based on the entropy coding scheme and the sub-block partition scheme. The partitioned sub-blocks are decoded based on the entropy coding scheme.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information.

Abstract: Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding/decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections.

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: Traditional audio encoders may conserve coding bit-rate by encoding fewer than all spectral coefficients, which can produce a blurry low-pass sound in the reconstruction. An audio encoder using wide-sense perceptual similarity improves the quality by encoding a perceptually similar version of the omitted spectral coefficients, represented as a scaled version of already coded spectrum. The omitted spectral coefficients are divided into a number of sub-bands. The sub-bands are encoded as two parameters: a scale factor, which may represent the energy in the band; and a shape parameter, which may represent a shape of the band. The shape parameter may be in the form of a motion vector pointing to a portion of the already coded spectrum, an index to a spectral shape in a fixed code-book, or a random noise vector. The encoding thus efficiently represents a scaled version of a similarly shaped portion of spectrum to be copied at decoding.

Abstract: Source signals, such as audio and/or video data, are encoded into multiple, consecutive frequency bands. These bands are referred to as coding layers. Rather than performing complex bit-slice operations, a disclosed technique enables an agile and simplified response to transmission channel throughput variations. Specifically, if it becomes necessary to restrict the rate of data transmission to avoid receiver buffer underflow resulting from transmission channel degradation, layers from the transmitted signal are omitted, beginning with the highest frequency bands. Efficient and agile bit rate scalability during data streaming through wired or wireless networks and during local playback is thus enabled.

Abstract: The techniques described are utilized for detection of noise and noise-like segments in audio coding. The techniques can include performing a prediction gain calculation, an energy compaction calculation, and a mean and variation energy calculation. Signal adaptive noise decisions can be made both in time and frequency dimensions. The techniques can be embodied as part of an AAC (advanced audio coding) encoder to detect noise and noise-like spectral bands. This detected information is transmitted in a bitstream using a signaling method defined for a perceptual noise substitution (PNS) encoding tool of the AAC encoder.

Abstract: A maximum of Nmax bits for encoding is defined for a set of parameters which may be calculated from a signal frame. The parameters for a first sub-set are calculated and encoded with N0 bits, where N0<Nmax. The allocation of Nmax?N0 encoding bits for the parameters of a second sub-set are determined and the encoding bits allocated to the parameters for the second sub-set are classified. The allocation and/or order of classification of the encoding bits are determined as a function of the encoding parameters for the first sub-set. For a total of N available bits for the encoding of the total parameters (N0<N=Nmax), the parameters for the second sub-set allocated the N?N0 encoding bits classified the first in said order are selected. Said selected parameters are calculated and encoded to give the N?N0 bits. The N0 encoding bits for the first sub-set and the N?N0 encoding bits for the selected parameters for the second sub-set are finally introduced into the output sequence of the encoder.

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 inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced.

Abstract: An encoder transforms at least a portion of a signal, counts the resulting transform coefficients having a zero value, and encodes the signal with the zero count. A decoder decodes the signal in order to recover the zero count. The decoder may also determine its own zero count of the signal as received and may compare the zero count that it determines to the recovered zero count. The decoder may be arranged to detect compression/decompression based upon results from the comparison, and/or the decoder may be arranged to prevent use of a device based upon results from the comparison.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: An audio encoder performs adaptive entropy encoding of audio data. For example, an audio encoder switches between variable dimension vector Huffman coding of direct levels of quantized audio data and run-level coding of run lengths and levels of quantized audio data. The encoder can use, for example, context-based arithmetic coding for coding run lengths and levels. The encoder can determine when to switch between coding modes by counting consecutive coefficients having a predominant value (e.g., zero). An audio decoder performs corresponding adaptive entropy decoding.

Abstract: A unified lossy and lossless audio compression scheme combines lossy and lossless audio compression within a same audio signal. This approach employs mixed lossless coding of a transition frame between lossy and lossless coding frames to produce seamless transitions. The mixed lossless coding performs a lapped transform and inverse lapped transform to produce an appropriately windowed and folded pseudo-time domain frame, which can then be losslessly coded. The mixed lossless coding also can be applied for frames that exhibit poor lossy compression performance.

Abstract: Presented herein is a reduced memory implementation technique of filterbank and block switching for real-time audio applications. Calculation of the pulse code modulated samples from the IMDCT samples and inverse window functions is simplified by exploiting the symetric qualities of the IMDCT function. As a result, memory requirements and operations are significantly reduced.

Abstract: Certain aspects of a method and system for a dual mode subband acoustic echo canceller with integrated noise suppression may include splitting an input signal into a lowband component and a highband component. The subbands of each of the lowband component and the highband component may be processed in order to reduce an echo associated with the input signal and to suppress the noise associated with the input signal.

Abstract: A method of and apparatus for parametric encoding and parametric decoding are provided. According to the method and apparatus, not all parameters for all component signals are generated and according to a time interval, parameters for some component signals are replaced by index information allowing similar previous time intervals to be found, thereby increasing encoding efficiency.