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 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: 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: The present invention proposes a device for controlling a stream of data packets, comprising: a buffer (2) adapted to receive a data packet stream from a data source (1) and to output said packet data stream with an output rate to a packet data communication network (7), a monitoring means (4) adapted to monitor a fill level of said buffer (2), a detection means (5) adapted to detect a fill level condition of said buffer (2), in which an incoming data packet has to be dropped, and adapted to control a dropping means (6) for dropping an incoming data packet upon detection of said fill level condition, and further comprising an output rate control means (5a, 8, 9, 10, 11) adapted to control an output rate adjusting means (3) which is adapted to adjust said output rate of said buffer (2), wherein said output rate control means, in response to said fill level condition detected by said detection means (5), issues a first control signal (5a) controlling said adjusting means (3) to increase a current output rate of

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: In general, this invention concerns speech encoding and decoding used in digital radio systems and a method by which the processing capacity required can be reduced in a telecommunication system using discontinuous transmission between a transmitter and receiver. In particular, the method according to the invention is used to match two telecommunication systems using different encoding methods between the transmitter and receiver. In the method, the signals transmitted by the transmitter are made suitable for the receiver in the signal path so that in the first step, at least one information parameter comprising at least two content identifiers is formed for each data frame of the data parameters (101) received. In the next step, data corresponding to the original data is synthesized from the data parameters (101) of the received frames, after which the synthesized data is transmitted for recoding with an encoding method suitable for the receiver.

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: 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: 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: A method provides a first network element (22) or a user equipment of a radio access system with a current value of at least one parameter (TS) during a change of access of the user equipment (20) from another network element (21) of a radio access system to the first network element (22). The parameter is required for creating headers for packets in which data of user equipment is transmitted. In order to guarantee continuous parameter values without jumps, the current parameter values (TS) are transmitted from the other network element (21), which knows the current parameter value, to the first network element (22) or to the user equipment, and that in addition, the network element (22) or the user equipment are synchronized to the other network element (21). The network element (22) or the user equipment then corrects the received parameter value (TS) based on the synchronization.

Abstract: A method and device perform error detection with partial checksum coverage by using a first transport protocol, wherein interface means are provided for requesting from a second lower-level transport protocol at least one lower-level header field to be subjected to checksum calculation over a predetermined portion of a data packet of the first transport protocol and the requested lower-level header fields. A checksum-based error processing function of the lower-level second transport protocol is disabled during the transmission of the data packet. Thereby, a checksum with partial coverage can be carried inside a higher-level data packet to provide Unequal Error Detection (UED) for error tolerant applications without requiring modifications of lower layer protocols.

Abstract: A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal uw(k). The time-warped reconstructed signal uw(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y{circumflex over ( )}w(k).

Abstract: The present invention proposes a device for controlling a stream of data packets, comprising: a buffer (2) adapted to receive a data packet stream from a data source (1) and to output said packet data stream with an output rate to a packet data communication network (7), a monitoring means (4) adapted to monitor a fill level of said buffer (2), a detection means (5) adapted to detect a fill level condition of said buffer (2), in which an incoming data packet has to be dropped, and adapted to control a dropping means (6) for dropping an incoming data packet upon detection of said fill level condition, and further comprising an output rate control means (5a, 8, 9, 10, 11) adapted to control an output rate adjusting means (3) which is adapted to adjust said output rate of said buffer (2), wherein said output rate control means, in response to said fill level condition detected by said detection means (5), issues a first control signal (5a) controlling said adjusting means (3) to increase a current output rate of

Abstract: The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals 13. In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server 12 receiving data packets from all terminals 13. Based on these data packets, then at least one terminal 13 currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals 13 which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals 13 participating in the conference call. The invention relates equally to a corresponding server and to a corresponding terminal.

Abstract: An error concealment method for multi-channel digital audio involves receiving an audio signal having audio data forming a first audio channel and a second audio channel included therein, wherein the first and second audio channels are correlated with each other in a manner so that a spatial sensation is typically perceived when listened to by a user. Erroneous first-channel data is detected in the first audio channel, and second-channel data is obtained from the second audio channel. The erroneous first-channel data of the first audio channel is corrected by using the second-channel data. Upon detection of the erroneous first-channel data, a spatially perceivable inter-channel relation between the first and second audio channels is determined, and the determined inter-channel relation is used when correcting the erroneous first-channel data of the first audio channel so as to preserve the spatial sensation perceived by the user.

Abstract: A method for transferring data over a communication link from a first unit to a second unit, the link comprising a first portion across which the data is carried by the transmission of data packets at regular intervals, and a second portion between the first portion and the first unit over which the data is carried in a form that is not synchronised with the transmission of data packets over the first portion; the method comprising transmitting to the first unit synchronisation information regarding the times at which packets are to be transmitted over the first portion.

Abstract: Speech data frames for transmitting control signalling messages are selected in accordance with the relative subjective importance of the speech signal data content of the frame. Speech frames are classified into frame types with lower priority frame types, such as non-speech frames, being selected first for the control message data and higher priority frame types, such as onset and transient, being avoided for selection due to the higher subjective contribution to speech quality.

Abstract: Circuitry, embodied in a media subsystem (10A), reproduces a speech or other type of audio signal, and is operable when playing back audio data for reducing synchronization delay. A method operates by, when a frame containing audio data is sent to a decoder (20), measuring the synchronization delay; determining by how much the synchronization delay should be adjusted; and adjusting the synchronization delay in a content-aware manner by adding or removing one or more audio samples in a selected current frame or in a selected subsequent frame so as not to significantly degrade the quality of the played back audio data. When the synchronization delay is adjusted by more than one audio sample, the adjustment can be made by all of the determined audio samples in one adjustment, or the adjustment is made by less than all of the determined audio samples by a plurality of adjustments. The step of adjusting selects, if possible, an unvoiced frame and discriminates against a transient frame.

Abstract: Apparatus, and an associated method, for facilitating synchronization of the operational codec modes of communication stations which communicate pursuant to a communication session. Signaling is provided to indicate allowed AMR (adaptive multi rate) codec modes to optimize speech connections in a GSM/EDGE radio access network (GERAN). RTP messages are defined, such as defining new message types or new field extensions in RTP messages to identify the codec modes.