Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: The present document relates to methods and apparatus for audio coding. In particular, the present document relates to methods and apparatus for enhanced block switching and/or bit allocation in audio coding of transient-tonal signals. A method of encoding samples of an audio signal comprises determining a first measure indicative of transient characteristics of the audio signal, determining a second measure indicative of tonal characteristics of the audio signal, selecting a transform length for the audio signal on the basis of the first measure and the second measure, and applying a time-frequency transform to a block of samples of the audio signal in accordance with the selected transform length, to thereby obtain a block of frequency coefficients corresponding to the block of samples of the audio signal.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal including spectral coefficients corresponding to frequencies up to a first cross-over frequency and performing parametric decoding at a second cross-over frequency to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method further includes extracting from the encoded audio bitstream a second waveform-coded signal including spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal. The interleaved signal is then combined with the first waveform-coded signal.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a multi-channel audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and discrete representation of the processed multi-channel audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: An arrangement of overlapping filter banks comprises a synthesis stage and an analysis stage. The synthesis stage receives a first signal segmented into time blocks and outputs, based thereon, an intermediate signal to be received by the analysis stage forming the basis for the computation of a second signal segmented into time frames. In an embodiment, the synthesis stage is operable to release an approximate value of the intermediate signal in a time block located L?1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of the first signal, so that the approximate value contributes, in the analysis stage, to the second signal. The delay is typically reduced by L?1 blocks. Applications include audio signal processing in general and real-to-complex conversion in particular.

Abstract: The present document relates to multichannel audio coding and more precisely to techniques for discrete multichannel audio encoding and decoding. In particular, the present document relates to systems and method for coding soundfields. An audio encoder (200) configured to encode a frame of a soundfield signal (110) comprising a plurality of audio signals is described. The audio encoder (200) comprises a transform determination unit (203, 204) configured to determine an energy-compacting orthogonal transform (V) based on the frame of the soundfield signal (110). Furthermore, the encoder (200) comprises a transform unit (202) configured to apply the energy-compacting orthogonal transform (V) to the frame of the soundfield signal (110), and configured to provide a frame of a rotated soundfield signal (112) comprising a plurality of rotated audio signals (E1, E2, E3).

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a multi-channel audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and discrete representation of the processed multi-channel audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: A device for generating a binaural signal based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having a virtual sound source position associated to each channel, is described. It includes a correlation reducer for differently processing, and thereby reducing a correlation between, at least one of a left and a right channel of the plurality of channels, a front and a rear channel of the plurality of channels, and a center and a non-center channel of the plurality of channels, in order to obtain an inter-similarity reduced set of channels; a plurality of directional filters, a first mixer for mixing outputs of the directional filters modeling the acoustic transmission to the first ear canal of the listener, and a second mixer for mixing outputs of the directional filters modeling the acoustic transmission to the second ear canal of the listener.

Abstract: The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.

Abstract: The present document relates to audio encoding/decoding. In particular, the present document relates to a method and system for improving the quality of encoded multi-channel audio signals. An audio encoder configured to encode a multi-channel audio signal according to a total available data-rate is described. The multi-channel audio signal is representable as a basic group (121) of channels for rendering the multi-channel audio signal in accordance to a basic channel configuration, and as an extension group (122) of channels, which—in combination with the basic group (122)—is for rendering the multi-channel audio signal in accordance to an extended channel configuration. The basic channel configuration and the extended channel configuration are different from one another.

Abstract: An arrangement of overlapping filter banks comprises a synthesis stage and an analysis stage. The synthesis stage receives a first signal segmented into time blocks and outputs, based thereon, an intermediate signal to be received by the analysis stage forming the basis for the computation of a second signal segmented into time frames. In an embodiment, the synthesis stage is operable to release an approximate value of the intermediate signal in a time block located L?1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of the first signal, so that the approximate value contributes, in the analysis stage, to the second signal. The delay is typically reduced by L?1 blocks. Applications include audio signal processing in general and real-to-complex conversion in particular.