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: An apparatus for efficiently browsing and selecting media content and reducing computational load of processing audio data may include a processor and a memory storing executable computer program code that causes the apparatus to at least perform operations including obtaining audio signals corresponding to items of media data. The audio data associated with the audio signals is played simultaneously. The computer program code may cause the apparatus to determine whether audio signals correspond to multi channel audio signals when determining whether to generate simplified audio signals. The computer program code may also cause the apparatus to determine directions/locations to output the audio data associated with the audio signals. The determined directions/locations correspond to directions that media data is currently being moved or locations of the media data. Corresponding computer program products and methods are also provided.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

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: Inputting of 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 including channel configuration information is shown along with deriving, from the channel configuration information, audio source location data describing at least one of horizontal and vertical positions of audio sources in the binaural audio signal; selecting, from a predetermined set of head-related transfer function filters, a left-right pair of head-related transfer function filters matching closest to the audio source location data, wherein the left-right pair of head-related transfer function filters is searched in a stepwise motion in a horizontal plane; and synthesizing a binaural audio signal from the at least one processed signal according to side information and the channel configuration information.

Abstract: A method comprising selecting a subset of audio sources from a plurality of audio sources, and transmitting signals from said selected subset of audio sources to an apparatus, wherein said subset of audio sources is selected in dependence on information provided by said apparatus.

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: 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: 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.

Abstract: A method for generating a parametrically encoded audio signal, the method comprising: inputting a multi-channel audio signal comprising a plurality of audio channels; generating at least one combined signal of the plurality of audio channels; and generating one or more corresponding sets of side information including channel configuration information for controlling audio source locations in a synthesis of a binaural audio signal.

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: 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 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 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: This invention relates to a method, a computer program product, apparatuses and a system for controlling a length of a frame buffer. The frame buffer is comprised in a receiver and buffers frames that are transmitted by a transmitter according to a frame transmission scheme and received at the receiver. The length of the frame buffer is controlled under consideration of a change in the frame transmission scheme.

Abstract: This invention relates to a method, a computer program product, apparatuses and a system for controlling a length of a frame buffer. The frame buffer is comprised in a receiver and buffers frames that are transmitted by a transmitter according to a frame transmission scheme and received at the receiver. The length of the frame buffer is controlled under consideration of a change in the frame transmission scheme.

Abstract: A system and method for efficiently implementing redundancy management in speech coding applications. According to various embodiments of the present invention, a sending device selects a redundant transmission level that is suitable for a current transmission channel condition while, at the same time, selecting the most suitable bit rate from the available codec mode set. With embodiments of the present invention, the redundant transmission level that is used is always optimal with regard to the selected codec modes, and no re-negotiation of the codec is needed.

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: Inputting of 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 including channel configuration information is shown along with deriving, from the channel configuration information, audio source location data describing at least one of horizontal and vertical positions of audio sources in the binaural audio signal; selecting, from a predetermined set of head-related transfer function filters, a left-right pair of head-related transfer function filters matching closest to the audio source location data, wherein the left-right pair of head-related transfer function filters is searched in a stepwise motion in a horizontal plane; and synthesizing a binaural audio signal from the at least one processed signal according to side information and the channel configuration information.