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: 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: Systems, methods and computer program codes are provided to facilitate error detection and timing synchronization of scalable data transmissions. To this end, checksum error detection is applied to the core layer and enhancement layers of the scalable payload data in such a way that dropping one or several enhancement layers from the payload data does not change the value of the checksum. Only one checksum is transmitted, e.g., in the payload or in the header of the lower-layer protocol. The transmitter modifies the encoded bit stream in such a manner that the entity in the network deploying the scalable payload and removing layers from the packet does not need to recalculate the checksum placed in the payload or packet header, even when the payload size is changed.

Abstract: A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value.

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: 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).

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.

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: A system and method for providing improved adaptive multi-rate wideband (AMR-WB) discontinuous transmission (DTX) synchronization. According to various embodiments, an indication on the start of the inactive speech period is signalled to the decoder via a voice activity detection (VAD) flag a predetermined number of frames before the DTX period will start, i.e., before the SID_FIRST frame is received. When the VAD flag indicates active speech, or when the VAD flag has been set to zero less than the predetermined number of frames ago, the received NO_DATA frame can be classified with a high degree of reliability as active speech, i.e., considered as transmitter, network or terminal-initiated signalling, and can be substituted by a SPEECH_LOST frame. When the VAD flag was set to zero eight frames ago or earlier, the NO_DATA frame is classified as DTX.

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 in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.

Abstract: Encoding audio signals for Discontinuous 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 quality of the selected encoding mode and encoding at least the active segments using the selected encoding mode that for a low quality encoding produce a lower number of “active” temporal section detections than for a high quality encoding mode, with comfort noise parameters producing less contrast from background noise for low quality encoding than for high quality modes.

Abstract: This invention relates to methods, a computer program product and apparatuses in the context of frame buffering. A buffering time for one or more frames received by a frame buffer is determined based on a specific buffering time associated with a specific frame and on information representative of a specific amount of data stored in the frame buffer.

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: A packet generator for generating packets from an input signal configured to: generate at least one first signal, dependent on the input signal, the first signal comprising a first relative time value; generate at least one second signal, dependent on the input signal and associated with the at least one first signal; and generate at least one indicator associated with each of the at least one second signal, each indicator dependent on the first relative time value.

Abstract: For supporting a decoding of encoded frames, which belong to a sequence of frames received via a packet switched network, it is detected whether a particular encoded frame has been received after a scheduled decoding time for the particular encoded frame and before a scheduled decoding time for a next encoded frame. In case the particular encoded frame is detected to have been received after its scheduled decoding time and before the scheduled decoding time for the next encoded frame, the particular encoded frame is re-scheduled to be decoded at the scheduled decoding time for the next encoded frame.

Abstract: For enhancing the performance of an adaptive jitter buffer, a desired amount of adjustment of a jitter buffer is determined at a first device using as a parameter an estimated delay. The delay comprises at least an end-to-end delay in at least one direction in a conversation. For this conversation, speech signals are transmitted in packets between the first device and a second device via a packet switched network. An adjustment of the jitter buffer is then performed based on the determined amount of adjustment.

Abstract: A method and related apparatus comprising: buffering an encoded audio input signal comprising at least one combined signal of a plurality of audio channels and one or more corresponding sets of side information parameters describing a multi-channel sound image; changing the length of at least one audio frame of said combined signal by adding or removing a segment of said combined signal; modifying said one or more sets of side information parameters with a change corresponding to the change in the length of said at least one audio frame of said combined signal; and transferring said at least one audio frame of said combined signal with a changed length and said modified one or more sets of side information parameters to a further processing unit.

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: Packets for a discontinuous transmission of a speech signal via a packet switched network may be provided in shorter transmission intervals during an active state and in longer transmission intervals during an inactive state. The active state may be selected whenever a speech signal comprises a speech burst, optionally with a hangover period after a respective speech burst. For enhancing the control of an adaptive jitter buffer at a receiver at the beginning of a respective transmission session, an active state is enforced in addition for a predetermined period at a beginning of a transmission session, irrespective of a presence of speech bursts. In case hangover periods are used, the length of the predetermined period exceeds the length of these hangover periods.

Abstract: A system and method for providing improved scalable error detection and cross-timing synchronization for packet-switched transmission. In one embodiment, checksum error detection is applied for the core layer and for enhancement layers of the scalable payload in such a way that dropping one or several enhancement layers from the payload does not change the value of the checksum. Only one checksum is transmitted, e.g., in the payload or in the header of the lower-layer protocol. The transmitter modifies the encoded bit stream in such a manner that the entity in the network deploying the scalable payload and removing layers from the packet does not need to recalculate the checksum placed in the payload or packet header, even when the payload size is changed. A prefix/tail bit field is added in the beginning/end of each enhancement layer to make the checksum check match with the common checksum.