Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: Biphase mark codes (BMC) are used in digital communications. Most BMC formats use preambles for rate determination and synchronization. A decoder compares the intervals of continuous high or continuous low voltages in a BMC stream to predetermined minimum and maximum values of half cell, full cell and one-and-a-half cell intervals for all supported sampling rates. If a pattern matching a preamble is found, the sampling rate is locked in and the decoder is synchronized to the BMC stream. Once locked, the decoder uses the predetermined minimum and maximum values at the locked rate to generate half cell, full cell and one-and-a-half cell indicators for a decoding state machine which decodes data in the BMC stream or decodes expected preambles.

Abstract: A method of determining a spatial coding mode for audio data sent by a sender entity (20) to a receiver terminal (10) adapted to receive said audio data in one or more sound reproduction formats. The method comprises the steps of the receiver terminal (10) indicating (1) to the sender entity (20) said reproduction format(s) in a given order of preference and the sender entity (20) determining a mode of spatial coding of the audio data compatible with a reproduction format taken in the order of preference and indicating (2) said reproduction format to the receiver terminal (10). Application to Voice over IP services.

Abstract: A method of processing a signal is disclosed, which includes receiving a downmix signal, object information and rendering information; and generating downmix processing information using the object information and the rendering information, the object information including modified object level information, wherein the modified object level information is determined using a level of a modified object generated by applying a downmix gain to an object. Accordingly, in an apparatus for processing a signal and method thereof according to the present invention, the signal is decoded using object information including modified object level information. Therefore, since a downmix gain is not transmitted, the present invention is able to reduce a used bit for object information and decode a signal using less information.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artifacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed.

Abstract: An audio matrix decoder is provided for decoding input signals Lt and Rt into output signals Lout, Cout, Sout, and Rout. The decoder circuit includes a main signal path having a passive matrix decoder and a cross talk canceller configured to apply variable gain values to intermediate signals and to compute the output signals Lout, Cout, Sout, and Rout. The decoder also includes a control signal path having a passive matrix decoder and a linear equation solver. The linear equation solver is configured to compute the variable gain values based on the intermediate signals and the necessary condition that re-encoded values of the output signals Lout, Cout, Sout, and Rout are equal to the input signals Lt and Rt. The linear equation solver is also configured to feed the variable gain values to the cross talk canceller for use in computing the output signals.

Abstract: Example embodiments allow for the creation, distribution, and use of flexible media formats. Example embodiments may allow individual content files to be rendered in multiple formats and versions. In addition, example embodiments may provide for granular rights management, which may allow users to access content files on a feature-by-feature basis.

Abstract: A broadcast transmitting apparatus and method, and a broadcast playback apparatus and method for providing an object-based audio by encoding and decoding a multichannel audio signal are provided. The broadcast transmitting apparatus may generate audio identification information used to determine whether the multichannel audio signal is an object-based audio signal. When the multichannel audio signal is determined to be the object-based audio signal, based on the audio identification information, the broadcast playback apparatus may control and output the multichannel audio signal for each channel.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal and generating a pseudo-surround signal using the downmix signal and the spatial information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal and a pseudo-surround decoding part generating a pseudo-surround signal from the downmix signal, using the spatial information.

Abstract: According to the invention, a device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; and a post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency domain representation of the second input channel and a complex prediction coefficient. The method comprises performing frequency-domain modifications selectively before or after upmixing.

Abstract: There is provided an apparatus for enabling karaoke. The apparatus includes a casing for the apparatus; a microphone array incorporated within the casing, the microphone array being concealed from a user by the casing; a controller coupled to the microphone array to at least process audio signals from the user input at the microphone array; a data storage device coupled to the controller, the data storage device being for storing pre-recorded songs used for karaoke, and for storing songs as sung by the user; and an image capturing device coupled to the controller, the image capturing device being for capturing images of the user while the user is singing, with the captured images of the user being stored on the data storage device.

Abstract: Methods and apparatus to extract data encoded in media content are disclosed. An example method includes sampling a media content signal to generate digital samples, determining a frequency domain representation of the digital samples, determining a first rank of a first frequency in the frequency domain representation, determining a second rank of a second frequency in the frequency domain representation, combining the first rank and the second rank with a set of ranks to create a combined set of ranks, comparing the combined set of ranks to a set of reference sequences, determining a data represented by the combined set of ranks based on the comparison, and storing the data in a memory device.

Abstract: An encoder for encoding an audio signal comprising at least two channels, the encoder configured to generate an encoded signal comprising at least a first part, a second part and a third part, wherein the encoder is further configured to: generate the first part of the encoded signal dependent on at least one combination of first and second channels of the at least two channels; generate the second part of the encoded signal dependent on at least one difference between the first and second channels of the at least two channels; and generate the third part of the encoded signal dependent on at least one energy ratio of the first and second channels of the at least two channels.

Abstract: According to the invention, a device (101, 101?) for postprocessing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver (103; 103?) for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal and a classification indication indicating a transient type of the at least one channel signal, wherein the classification indication is associated to the at least one channel signal, and a postprocessor (105; 105?) for postprocessing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication.

Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.

Abstract: A method of decoding an audio signal is disclosed, The present invention includes the steps of receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal, extracting the ambient component signal and the source component signal of each of the channels based on correlation between the channel signals, modifying the ambient component signal using surround effect information, and generating the audio signal including a plurality of channels using the modified ambient component signal and the source component signal. Accordingly, in an apparatus for decoding an audio signal and method thereof according to the present invention, an ambient component signal is extracted and modified based on correlation and the modified ambient and source component signals are outputted using different signal output units, respectively. Therefore, the present invention enhances a stereo effect of the audio signal.

Abstract: A method and apparatus for decoding a multichannel audio signal comprising the steps of receiving a downmix audio signal and an interchannel cross correlation parameter; deriving an interchannel phase difference parameter from the received interchannel cross correlation parameter; and calculating a decoded multichannel audio signal for the received downmix audio signal depending on the derived interchannel phase difference parameter.

Abstract: The invention relates to downmixing techniques by which output audio signals are obtained from input audio signals partitioned into subgroups. A variable common gain limiting factor is applied to all downmix coefficients that govern the contributions from the input signals in a subgroup. While preserving the proportions between signal values within a subgroup, the invention makes it possible to limit the gain of different input signal subgroups to different extents, so that relatively more perceptible signals can be limited relatively less. It then becomes possible to achieve a consistent dialogue level while transitioning in a less perceptible fashion between signal portions with and without gain limiting. Embodiments of the invention include a method, a mixing system and a computer-program product.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating source mapping information corresponding to each source of multi-sources by using spatial information indicating features between the multi-sources; generating at least one rendering information by using the source mapping information and filter information having a surround effect; and performing interpolation by using neighbor rendering information of the at least one rendering information.

Abstract: For a multi-channel reconstruction of audio signals based on at least one base channel, an energy measure is used for compensating energy losses due to an predictive upmix. The energy measure can be applied in the encoder or the decoder. Furthermore, a decorrelated signal is added to output channels generated by an energy-loss introducing upmix procedure. The energy of the decorrelated signal is smaller than or equal to an energy error introduced by the predictive upmix. Thus, problems occurring for prediction based up-mix methods such as up-mixing signals that are coded with High Frequency Reconstruction techniques are solved, so that the correct correlation between the up-mixed channels is obtained or the up-mix is adapted to arbitrary down-mixes.

Abstract: Methods, systems, and apparatus are presented for decoding an audio signal that includes bandwidth extension data. An audio signal that includes core audio data and bandwidth extension data can be received in a decoder. The core audio data can be associated with a core portion of an audio signal, such as the frequency range below a cutoff frequency, and the bandwidth extension data can be associated with an extended portion of the audio signal, such as a frequency range above the cutoff frequency. The core audio data can be decoded to generate a decoded core audio signal in a time domain representation. Further, an extended portion of the audio signal can be reconstructed in accordance with extension data and decoded core audio signal. Additionally, the decoded core audio signal can be lowpass filtered and the extended portion can be highpass filtered before being combined to generate a decoded output signal.

Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

Abstract: The present invention relates to an encoding device and method, and a decoding device and method, and a program which enable music signals to be played with higher sound quality by expanding a frequency band. A band pass filter divides an input signal into multiple subband signals, a feature amount calculating circuit calculates feature amount using at least any one of the divided multiple subband signals and the input signal, a high-frequency subband power estimating circuit calculates an estimated value of high-frequency subband power based on the calculated feature amount, and a high-frequency signal generating circuit generates a high-frequency signal component based on the multiple subband signals divided by the band pass filter and the estimated value of the high-frequency subband power calculated by the high-frequency subband power estimating circuit.

Abstract: The present invention relates to a signal processing apparatus and a signal processing method, an encoder and an encoding method, a decoder and a decoding method, and a program capable of reproducing music signal having a better sound quality by expansion of frequency band. A sampling frequency conversion unit converts a sampling frequency of an input signal, and a sub-band division circuit divides the input signal after the sampling conversion into sub-band signals of sub-bands having the number corresponding to the sampling frequency. A pseudo high band sub-band power calculation circuit calculates pseudo high band sub-band powers based on low band signals of the input signal and coefficient tables having coefficients for the respective high band sub-bands. A pseudo high band sub-band power difference calculation circuit compares high band sub-band powers and the pseudo high band sub-band powers to each other and selects a coefficient table from plural coefficient tables.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and spatial information regarding a plurality of channels from an input bitstream, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the down-mix signal using the spatial information and a filter, wherein the sum of the number of valid signals of the down-mix signal, the number of valid signals of the spatial information, and the number of valid signals of co-efficients of the filter is less than the number of valid signals of the 3D down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: A multi-channel audio signal decoding method and device is provided. The multi-channel audio signal decoding method includes receiving a first multi-channel audio signal; performing a first decoding procedure on the first multi-channel audio signal to generate a second multi-channel audio signal; performing a second decoding procedure on a first single-channel audio data of the second multi-channel audio signal to generate a first single-channel audio signal when the first single-channel audio data belongs to a first classification; and performing a third decoding procedure on a second single-channel audio data of the second multi-channel audio signal to generate a second single-channel audio signal when the second single-channel audio data belongs to a second classification. The number of instructions of the third decoding procedure is less than that of the second decoding procedure.

Abstract: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: A method for simulating spatially extended sound sources comprising: panning a first input signal over a plurality of output channels to generate a first multi-channel directionally encoded signal; panning a second input signal over the plurality of output channels to generate a second multi-channel directionally encoded signal; combining the first and second multi-channel directionally encoded signals to generate a plurality of loudspeaker output channels; and applying a bank of decorrelation filters on the loudspeaker output channels.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal and a downmix processing information; and, processing the downmix signal using a downmix processing information, comprising: de-correlating the downmix signal; and, mixing the downmix signal and the de-correlated signal in order to output the processed downmix signal, wherein the downmix processing information is estimated based on an object information and a mix information is disclosed.

Abstract: This audio matrix surround decoder requires minimal digital processing, useful in portable applications, particularly in playback from a portable player using a headphone or loudspeaker virtualizer. In one embodiment it pans inputs Lt and Rt to outputs associated with front directions in response to a measure of the sum of Lt and Rt being greater than a measure of the difference between Lt and Rt, and pans Lt and Rt to outputs associated with rear directions in response to a measure of the sum of Lt and Rt being less than a measure of the difference between Lt and Rt. Lt and Rt are modified to shift the direction of reproduced signals.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating source mapping information corresponding to each source of multi-sources by using spatial information indicating features between the multi-sources; generating sub-rendering in formation by applying filter information giving a surround effect to the source mapping in formation per the source; generating rendering information for generating a surround signal by integrating the at least one of the sub-rendering information; and generating the surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: Provided is a system and method for front wave field synthesis (WFS) to provide a surround sound with a reduced number of loudspeakers. An apparatus for encoding a front WFS signal in the front WFS system may include a content receiving unit to receive WFS content including the first channels corresponding to a front array speaker and the second channels corresponding to a rear surround sound speaker, and an encoding unit to encode the first channels and the second channels using a multichannel encoder corresponding to the number of the channels included in the WFS content.

Abstract: An audio information processing apparatus for processing input audio information is adapted to input an audio signal composed of front area audio information that should be input to a front speaker and rear area audio information that should be input to a plurality of speakers. A mixer is adapted to mix the rear area audio information in accordance with an instruction from an instruction circuit.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving downmix information, object information and mix information, generating and transferring multi-channel information using at least one of the downmix information, the object information and the mix information, and selectively generating and transferring either first gain information or extra multi-channel information including second gain information in accordance with a decoding mode using at least one of the object information and the mix information.

Abstract: An apparatus and method for encoding/decoding a multi-channel signal may be provided. The apparatus of encoding a multi-channel signal may insert information about whether to encode a phase parameter indicating phase information of a plurality of channels, included in the multi-channel signal, in a bitstream of the multi-channel signal. The apparatus of decoding a multi-channel signal may determine whether to up-mix a mono signal using the phase parameter based on the information about whether to encode.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, a first multi-channel information, and an object information; processing the downmix signal using the object information and a mix information; and, transmitting one of the first multi-channel information and a second multi-channel information according to the mix information, wherein the second channel information is generated using the object information and the mix information is disclosed.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: The present invention includes an audio signal receiving unit receiving the audio signal having a plurality of channel signals including an ambient component signal and a source component signal; an ambient component signal extracting unit extracting the ambient component signal of each of the channels based on correlation between the channel signals; an ambient component signal modifying unit modifying the ambient component signal using surround effect information; a source component signal extracting unit extracting the source component signal of each of the channels based on the correlation between the channel signals; a first signal output unit outputting the modified ambient component signal and the source component signal; and a second signal output unit outputting the audio signal or the source component signal.

Abstract: An audio-signal processing device includes a decoding unit that decodes a compressed audio stream to obtain audio signals for a predetermined number of channels; a signal processing unit that generates 2-channel audio signals including left-channel audio signals and right-channel audio signals, on the basis of the predetermined-number-of-channels audio signals; and a coefficient setting unit that sets filter coefficients corresponding to the impulse responses for the digital filters, on the basis of format information of the compressed audio stream. The signal processing unit uses digital filters to convolve impulse responses for paths from sound-source positions of the channels to the left and right ears of a listener with the corresponding predetermined-number-of-channels audio signals and adds corresponding results of the convolutions for the channels to generate the left-channel audio signals and the right-channel audio signals.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of generating rendering information by using spatial information indicating features between multi-sources and filter information having a surround effect; and generating a surround signal by applying the rendering information to a downmix signal generated by downmixing the multi-sources, wherein a tab number of a filter used in applying the rendering information to generate the surround signal is variable.

Abstract: An encoding apparatus includes a down-mixing unit for down-mixing first and second input signals, and controlling a spatial parameter comprising or not comprising a first parameter which indicates a correlation between the first and second input signals to be transmitted to a decoding apparatus according to whether any one of the first and second input signals has a magnitude that is equal to or less than a predetermined threshold value; and a transmitting unit for transmitting the spatial parameter to the decoding apparatus.

Abstract: A speech enhancement device and a method for the same are included. The device includes a down-converter, a speech enhancement processor, and an up-converter. The method includes steps of down-converting audio signals to generate down-converted audio signals; performing speech enhancement on the down-converted audio signals to generate speech-enhanced audio signals; and up-converting the speech enhancement audio signals to generate up-converted audio signals.

Abstract: The present invention provides a frequency-domain spatial audio coding framework based on the perceived spatial audio scene rather than on the channel content. In one embodiment, time-frequency spatial direction vectors are used as cues to describe the input audio scene.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0?k<Na). A band extension section extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). An extended spectrum assignment section assigns extended spectrum S1?(k)(Na?k<Nb) input to the extended frequency band from the outside. A spectral information specification section outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from the extended spectrum assignment section as a code.

Abstract: A motion picture soundtrack reproduction system has a center front soundtrack channel and a plurality of other soundtrack channels. A volume control adjusts the gain of all the channels. The volume control has a range of settings from a minimum to a maximum, the gain of the center front channel having substantially a first relationship to the volume control settings and the gain of the other channels having substantially a second relationship to the volume control settings, the relationships being such that for a range of volume control settings less than a first setting the gain of the center front channel remains substantially constant while the gain of the other channels decreases as the setting decreases or decreases more gradually than the gain of the other channels as the setting decreases.