Abstract: A method, a chipset, a receiver, a transmitter, an electronic device and a system for enabling a control of jitter management of an audio signal is described, wherein the audio signal is distributed to a sequence of frames that are received via a packet switched network, the received frames comprising active audio frames and non-active audio frames, wherein a concatenation of subsequent active audio frames represents an active audio burst, wherein a discrete information of audio activity of the audio signal via the packet switched network is received, the end of an active audio burst is determined based on the received discrete information of audio activity, and wherein jitter compensation of the received frames is controlled on the basis of the determined end of an active audio burst. The invention further relates to a corresponding software program product storing a software code for controlling jitter management of an audio signal.

Abstract: Encoding audio signals with selecting an encoding mode for encoding the signal categorizing the signal into active segments having voice activity and non-active segments having substantially no voice activity by using categorization parameters depending on the selected encoding mode and encoding at least the active segments using the selected encoding mode.

Abstract: The invention relates to a method of synthesizing a mono audio signal 3 based on an available encoded multichannel audio signal 2. The encoded multichannel audio signal 2 is assumed to comprise at least for a part of an audio frequency band separate parameter values for each channel of the multichannel audio signal. In order to reduce the processing load in synthesizing the mono audio signal 2, it is proposed that the parameter values of the multiple channels are combined at least for a part of an audio frequency band in the parameter domain. The combined parameter values are then used for synthesizing the mono audio signal. The invention relates equally to a corresponding audio decoder, to a corresponding coding system and to a corresponding software program product.

Abstract: For time-scaling an audio signal, which is distributed to a sequence of frames, one scaling period is removed from the audio signal within a current frame, in case the audio signal is to be shortened in the time-scaling. Moreover, a segment of the audio signal following upon the removed scaling period is modified, for concealing said removal of a scaling period, at least partly in a subsequent frame, in case a segment of the audio signal following upon the removed scaling period within the current frame is shorter than desired for the modification.

Abstract: For time-scaling an audio signal that is distributed to a sequence of frames, frames of the sequence of frames are time scaled whenever needed, resulting in a sequence of variable sized frames. An audio signal in the sequence of variable sized frames is then re-divided into a sequence of equal sized frames for further processing.

Abstract: For controlling a time-scaling of an audio signal, the audio signal being distributed to a sequence of frames that are received via a packet switched network, a change in a delay of received frames is detected. Moreover, an amount of time scaling that is to be applied to received frames for compensating for the detected change is determined. Further, a kind of the change is determined. Further, a length of a time window within which a time scaling of the determined amount is to be completed is determined depending on the determined kind of the change.

Abstract: A method for synthesizing a binaural audio signal, the method comprising: inputting a parametrically encoded audio signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information describing a multi-channel sound image; and applying a predetermined set of head-related transfer function filters to the at least one combined signal in proportion determined by the corresponding set of side information to synthesize a binaural audio signal. A corresponding parametric audio decoder, parametric audio encoder, computer program product, and apparatus for synthesizing a binaural audio signal are also described.

Abstract: A method for synthesizing a binaural audio signal, the method comprising: inputting a parametrically encoded audio signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information describing a multi-channel sound image; and applying a predetermined set of head-related transfer function filters to the at least one combined signal in proportion determined by said corresponding set of side information to synthesize a binaural audio signal.

Abstract: A method and a device for source coding with a time advanced excitation signal. During an encoding process, a source data signal is first divided into consecutive blocks, then a first set of parameters related to a filter describing properties of a first block covering a first time period is extracted, followed by the extraction of a second set of parameters related to an excitation signal for said filter, where said second set of parameters is determined from and describing properties of both the first block and a second block following the first block within a second time period starting later than said first time period and extending outside said first time period.

Abstract: The present invention provides a method for compensating transient effects in transform coding and decoding of a combined speech and audio in electronic devices by using a transform based time-frequency domain codec. The method can combine, e.g., a CELP (code excited linear prediction) type speech codec and a transform type audio codec. The invention describes a compensation method to handle the transient (e.g., from the CELP coding to the transform coding) in transform coding when the number of quantized transform coding coefficients is lower than in the output of the transform.

Abstract: A scalable encoder (100) for encoding a media signal is described. It comprises first encoding means (210) for producing a first data stream (102), which is a core data stream relating to the media signal (101), having a first bit-rate; second encoding means (230) for producing a second data stream (103), which comprises a set of enhancement data streams relating to the media signal, having a second bit-rate; and a multiplexer (110) for combining at least the first data stream and the second data stream into a third data stream (104). The scalable encoder it further comprises control means (420, 421, 422), which is arranged to receive control information (401), to determine a target combination of the first data stream and the second data stream in the third data stream according to the control information and to adjust the combination of the first data stream and the second data stream in the third data stream by affecting the first and the second bit-rates.

Abstract: A first mobile station generates an uplink error check code for a data frame for uplink transmission, and transmits the data frame and its associated uplink error check code to a cellular network. The network determines the quality of the received data frame and generates a downlink error check code for the received data frame. The received frame quality is determined as good or bad on the basis of a function of the received data or a parameter indicative of the quality of the received frame quality and the received data respectively. The data frame and the associated downlink error check code is transmitted to the second mobile station. The second mobile station decodes the received data frame and the associated downlink error check code and determines if the received data frame and the associated downlink error check code represents a good quality frame.

Abstract: The invention relates to enabling a compensation of packet losses in a packet based transmission of data frames, wherein packets provided for transmission include a first type of frames corresponding to a respective data frame encoded using a first bit rate coding mode and a second type of frames corresponding to a respective data frame encoded using a second bit rate coding mode. In order to limit the processing power in the packet generation, parameters are extracted from a data frame which is to be transmitted in accordance with the first bit rate coding mode. The extracted parameters are quantized in accordance with the first bit rate coding mode to obtain quantized parameters forming a frame of the first type. In addition, a frame of the second type is generated based on the parameters extracted for the frame of the first type and/or on the quantized parameters of the frame of the first type.

Abstract: A method for supporting an encoding of an audio signal is shown, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module 2,3 in which this method is implemented, to a device 1 and a system comprising such a module 2,3, and to a software program product including a software code for realizing the proposed method.

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: An encoder comprises an input for inputting frames of an audio signal in a frequency band, an analysis filter dividing the frequency band into lower and higher frequency bands, a first encoding block for encoding the audio signals of the lower frequency band, a second encoding block for encoding the audio signals of the higher frequency band, and a mode selector for selecting an operating mode for the encoder among at least a first mode where signals only on the lower frequency band are encoded, and a second mode where signals on both the lower and higher frequency band are encoded. The encoder has a scaler to gradually change the encoding properties of the second encoding block in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.

Abstract: An encoder comprising an input for inputting frames of an audio signal in a frequency band, at least a first excitation block for performing a first excitation for a speech like audio signal, and a second excitation block for performing a second excitation for a non-speech like audio signal. The encoder further comprises a filter for dividing the frequency band into a plurality of sub bands each having a narrower bandwidth than the frequency band. The encoder also comprises an excitation selection block for selecting one excitation block among the at least first excitation block and the second excitation block for performing the excitation for a frame of the audio signal on the basis of the properties of the audio signal at least at one of the sub bands. The invention also relates to a device, a system, a method and a storage medium for a computer program.

Abstract: Methods and arrangements are disclosed for digitally encoding and decoding sound—An input signal is split (811) into a primary frequency band and at least one secondary frequency band. The parts of the input signal in the frequency bands are separately encoded. Certain characteristics of the input signal in the primary frequency band and corresponding characteristics of the input signal in at least one secondary frequency band are examined (302, 303, 814) in order to find out, whether there is certain resemblance therebetween. Alternatively certain characteristic features of the process applied to encoding the primary frequency band extracted (305, 813) and used (307) in encoding the secondary frequency band, or such extracted characteristic features are replaced (306, 501, 701, 815) with a locally generated, independent set of corresponding features.

Abstract: The present invention describes a novel methodology for adjusting a bandwidth extension algorithm by adapting one or more of enhancing perception parameters (e.g., a signal level, a signal energy and/or a gain) of a high-band encoded signal based on the characteristics of the input signal and an encoding performance in a low band with a codec utilizing audio-band-split coding by separate encoders and decoders for each audio band. The adaptation is based on the low-band coding algorithm. It can be at least two types of such an algorithm: e.g., an algebraic code excitation linear prediction (ACELP) algorithm for a speech-like input signal and a transform algorithm of a non-speech-like input signal, such that when the ACELP coding is selected, the corresponding enhancing perception parameter is gradually tuned down and when the encoding algorithm is changed to the transform coding, the corresponding enhancing perception parameter is gradually tuned up.

Abstract: A method for performing variable rate speech coding in the speech codec comprising a plurality of speech codec modes operating at different bit rates, the speech encoded by said speech codec being arranged for transmission in a telecommunications network. Information on an active speech codec mode set to be supported is received from the telecommunications network, in response to which the supported speech codec modes that correspond to the active codec mode set determined in the telecommunications network will be activated. Thereafter, speech signals to be applied to the speech codec are encoded with the activated speech codec modes such that the speech codec mode of the substantially lowest bit rate is adapted to the speech frames comprised by the speech signals such that in view of the channel conditions in the telecommunications network the level of residual error in coding will be substantially minimized at the same time.