Abstract: In a method of audio processing, a plurality of audio samples are received, and are concatenated to form a composite audio signal. The composite audio signal is analysed to identify audio artefacts associated with concatenation in the composite audio signal, and any identified audio artefacts are compensated for, to form a corrected composite audio signal. The corrected composite audio signal is provided to a voice biometrics module.

Abstract: There is provided a sound source separation method of carrying out sound source separation of an audio signal inputted from an input device by using a modeled sound source distribution, by an information processing apparatus provided with a processing device, a storage device, the input device, and an output device. In this method, as a condition followed by the model, sound sources are independent of one another, powers which the sound sources have are modeled for each of frequency bands obtained through band division, a relationship among the powers for the frequency bands different from each other is modeled by nonnegative matrix factorization, and components obtained through the division of the sound source follow a complex normal distribution.

Abstract: An audio post-processor for post-processing an audio signal having a time-variable high frequency gain information as side information includes: a band extractor for extracting a high frequency band of the audio signal and a low frequency band of the audio signal; a high band processor for performing a time-variable modification of the high frequency band in accordance with the time-variable high frequency gain information to obtain a processed high frequency band; and a combiner for combining the processed high frequency band and the low frequency band. Furthermore, a pre-processor is illustrated.

Abstract: An apparatus includes a receiver configured to receive at least one encoded signal that includes inter-channel bandwidth extension (BWE) parameters. The device includes a decoder configured to generate a mid-channel time-domain high-band signal by performing bandwidth extension based on the at least one encoded signal. The decoder is configured to generate, based on the mid-channel time-domain high-band signal and the inter-channel BWE parameters, a first channel time-domain high-band signal and a second channel time-domain high-band signal. The first channel time-domain high-band signal is selectively based on an adjustment spectral shape parameter responsive to whether the inter-channel BWE parameters include an adjustment spectral shape parameter. The decoder is configured to generate a target channel signal based at least in part on the first channel time-domain high-band signal, and to generate a reference channel signal based at least in part on the second channel time-domain high-band signal.

Abstract: A decoder for generating an audio output signal having one or more audio output channels is provided, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal. Moreover, an encoder is provided.

Abstract: The lossless coding method includes selecting one of a first coding method and a second coding method, based on a range in which a quantization index of energy is represented, and coding the quantization index by using the selected coding method. The lossless decoding method includes determining a coding method of a differential quantization index of energy included in a bitstream and decoding the differential quantization index by using one of a first decoding method and a second decoding method based on a range in which a quantization index of energy is represented, in response to the determined coding method.

Abstract: An audio signal processing method and an audio equalizer, both implemented in an embedded system without incurring excessive computation, use a Kaiser-Bessel-derived (KBD) window and an Overlap-and-Add (OLA) processing to eliminate signal distortion of a time domain audio signal during signal conversion and to generate filters according to audio effects desired by a user.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: A method, device and integrated circuit for receiving signals in a narrow bandwidth range from a base station of a network, the narrow bandwidth range being a portion of an overall bandwidth range of the network and the signals comprising one or more of cell-specific reference signal (CRS) tones, determining whether a condition exists based on the one or more CRS tones, when the condition exists, extending the narrow bandwidth range to an extended narrow bandwidth range, the extended narrow bandwidth range including at least one further CRS tone than the narrow bandwidth range and monitoring the extended narrow bandwidth range to receive further signals comprising the at least one further CRS tone.

Abstract: Methods and apparatuses are provided for performing end-to-end speech recognition training performed by at least one processor. The method includes receiving, by the at least one processor, one or more input speech frames, generating, by the at least one processor, a sequence of encoder hidden states by transforming the input speech frames, computing, by the at least one processor, attention weights based on each of the sequence of encoder hidden states and a current decoder hidden state, performing, by the at least one processor, a decoding operation based on a previous embedded label prediction information and a previous attentional hidden state information generated based on the attention weights; and generating a current embedded label prediction information based on a result of the decoding operation and the attention weights.

Abstract: The disclosure relates to a method for transmitting an item of information in a communication channel linking a transmitter to a recipient, the method including the following successive steps: the determination, by the transmitter, of a first intelligible sound signal representative of the item of information to be transmitted, the acoustic processing, by the transmitter, of the first sound signal so as to produce a second sound signal unintelligible to the recipient, the transmission of the second sound signal from the transmitter to the recipient, the transmission of the first sound signal from the transmitter to the recipient, the transmission of the first sound signal being triggered by the transmission of the second sound signal, the second transmission of the second sound signal from the transmitter to the recipient, the second transmission of the second sound signal being triggered by the transmission of the first signal.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: A downmixer for downmixing at least two channels of a multichannel signal having the two or more channels includes: a processor for calculating a partial downmix signal from the at least two channels; a complementary signal calculator for calculating a complementary signal from the multichannel signal, the complementary signal being different from the partial downmix signal; and an adder for adding the partial downmix signal and the complementary signal to obtain a downmix signal of the multichannel signal.

Abstract: A broadcast transmitting apparatus and method, and a broadcast playback apparatus and method for providing an object-based audio by encoding and decoding a multichannel audio signal are provided. The broadcast transmitting apparatus may generate audio identification information used to determine whether the multichannel audio signal is an object-based audio signal. When the multichannel audio signal is determined to be the object-based audio signal, based on the audio identification information, the broadcast playback apparatus may control and output the multichannel audio signal for each channel.

Abstract: The present invention relates to a method and an apparatus for encoding and decoding spectrum coefficients in the frequency domain. The spectrum encoding method may comprise the steps of: selecting an encoding type on the basis of bit allocation information of respective bands; performing zero encoding with respect to a zero band; and encoding information of selected significant frequency components with respect to respective non-zero bands. The spectrum encoding method enables encoding and decoding of spectrum coefficients which is adaptive to various bit-rates and various sub-band sizes. In addition, a spectrum can be encoded using a TCQ method at a fixed bit rate using a bit-rate control module in a codec that supports multiple rates. Encoding performance of the codec can be maximised by encoding high performance TCQ at a precise target bit rate.

Abstract: A method and a system for correcting energy distributions of audio signal are proposed. The method is applicable to a head-mounted device having a motion sensor, a left speaker, and a right speaker and includes the following steps. A rotation angle of the head-mounted device is detected by the motion sensor. Dual-channel audio signals corresponding to the left and right speakers are obtained. The dual-channel audio signals are converted to multi-channel audio signals with the number of channels greater than or equal to 5. Four acoustic source positions of the left and right speakers are defined to convert the multi-channel audio signals to four-channel audio signals of the left and right speakers. Energy distributions of the four-channel audio signals of the left and right speakers are corrected according to the rotation angle and the four acoustic source positions to respectively generate a left output signal and a right output signal.

Abstract: According to examples, an apparatus may include a processor and a memory on which is stored machine readable instructions that are to cause the processor to determine a reference frame from a plurality of frames received at multiple different times, in which each of the plurality of frames includes audio signal data, and in which the reference frame includes audio signal data that identifies a highest audio signal level among audio signals identified in the plurality of frames. The reference frame may be time-aligned with each of the plurality of frames other than the reference frame to obtain respective time-aligned frames. The audio signals identified in each of the respective time-aligned frames may be added together to generate respective added audio signals. The respective added audio signals may be combined together to obtain a combined audio signal and the combined audio signal may be outputted.

Abstract: Provided are a terminal device and method of performing a call function transmitting ambient audio with high sensitivity. A terminal device performing a call function with at least one external device via a network may include a receiver configured to receive at least one of an audio transmission signal and a video transmission signal to be transmitted to the external device; a processor configured to analyze at least one of the audio transmission signal and the video transmission signal, select one of a speech mode and an audio mode, based on a result of the analysis, and compress the audio transmission signal, based on the selected mode; a communicator configured to transmit the compressed audio transmission signal to the external device, and receive an audio reception signal from the external device; and an output unit configured to output the audio reception signal.

Abstract: A signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal. An envelope information generating unit generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. The high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream.

Abstract: A sound effect configuration method and a mobile terminal are provided. The method comprises: parameters of sound effect are loaded in a running memory of the mobile terminal; responsive to detection of a playing control instruction for an audio stream of a target application, it is determined whether the parameters of sound effect loaded in the running memory comprise a parameter of sound effect corresponding to the target application; responsive to determining that the parameters of sound effect loaded in the running memory comprise the parameter of sound effect corresponding to the target application, the parameter of sound effect corresponding to the target application is read from the running memory; and the audio stream of the target application is configured by virtue of the parameter of sound effect.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

Abstract: Embodiments of the present application provide a coding/decoding method, apparatus, and system. According to the coding method, de-emphasis processing is performed on a full band signal by using a de-emphasis parameter determined according to a characteristic factor of an input audio signal, and then the full band signal is coded and sent to a decoder, so that the decoder performs corresponding de-emphasis decoding processing on the full band signal according to the characteristic factor of the input audio signal and restores the input audio signal. This resolves a prior-art problem that an audio signal restored by a decoder is apt to have signal distortion, and implements adaptive de-emphasis processing on the full band signal according to the characteristic factor of the audio signal to enhance coding performance, so that the input audio signal restored by the decoder has relatively high fidelity and is closer to an original signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: An apparatus for phase reconstruction from a magnitude spectrogram of an audio signal is provided. The apparatus includes a frequency change determiner being configured to determine a change of a frequency for each time-frequency bin of a plurality of time-frequency bins of the magnitude spectrogram of the audio signal depending on the magnitude spectrogram of the audio signal, and a phase reconstructor being configured to generate phase values for the plurality of time-frequency bins depending on the changes of the frequencies determined for the plurality of the time-frequency bins.

Abstract: A system for converting a channel-based 3D audio signal to a higher-order Ambisonics HOA audio signal, the channel-based 3D audio signal is transformed from time domain to frequency domain. A primary ambient decomposition is carried out for three-channel triplets of blocks of the domain channel-based 3D audio signal, wherein directional signals and ambient signals are provided for each triplet. From the directional signals directional information of a total directional signal for each triple is derived. That total directional signal is HOA encoded according to the derived directions, and ambient signals are HOA encoded according to channel positions. The HOA coefficients of the HOA encoded directional signal and the HOA coefficients of the HOA encoded ambient signal are superimposed in order to obtain a HOA coefficients signal for the channel-based 3D audio signal, followed by a transformation into time domain.

Abstract: In an audio signal decoding method, a decoded frequency domain signal of a current frame of the audio signal is obtained by decoding a received bitstream; a predicted frequency domain signal of the current frame is obtained according to the decoded frequency domain signal the current frame when the decoded frequency domain signal meets anyone of two given conditions; and a time domain signal of the current frame is obtained according to the decoded frequency domain signal and the predicted frequency domain signal.

Abstract: Audio video synchronization and alignment or alignment of audio to some other external clock are rendered more effective or easier by treating fragment grid and frame grid as independent values, but, nevertheless, for each fragment the frame grid is aligned to the respective fragment's beginning. A compression effectiveness lost may be kept low when appropriately selecting the fragment size. On the other hand, the alignment of the frame grid with respect to the fragments' beginnings allows for an easy and fragment-synchronized way of handling the fragments in connection with, for example, parallel audio video streaming, bitrate adaptive streaming or the like.

Abstract: An example apparatus includes a memory configured to store the audio data; and one or more processors in communication with the memory, the one or more processors configured to: decode, from an encoded audio bitstream, a unique identifier for each of a plurality of subbands of audio data; perform inverse pyramid vector quantization (PVQ) using a compact map to reconstruct a residual vector for each subband of the plurality of subbands of the audio data based on the unique identifier for the respective subband of the plurality of subbands of the audio data, wherein the compact map is generated using structural unification of vectors across subbands and relational compression, and wherein the unique identifiers correspond to codevectors; and reconstruct, based on the residual vectors and energy scalars for each subband, the plurality of subbands of the audio data.

Abstract: A waveform monitor device, or media analysis device, to monitor a video signal. The waveform monitor device may include an input to receive the video signal, the video signal having a plurality of frames, a memory to store the received video signal, a processor coupled to the memory, and a display. The processor separates the video signal into at least two component signals, for at least one component signal, determines a peak value of the at least one component signal for at least one frame of the plurality of frames, generates a marker at the peak value, determines if the peak value violates a predetermined threshold, and generates an alert when the peak value violates the predetermined threshold. The at least one component signal, the marker at the peak value on the component signal and the alert when the peak value violates the predetermined threshold are displayed on the display to allow a user to quickly determine if a video signal is within a required threshold.

Abstract: There is presented mechanisms for handling input envelope representation coefficients. A method is performed by an encoder of a communication system. The method comprises determining envelope representation residual coefficients as first compressed envelope representation coefficients subtracted from the input envelope representation coefficients. The method comprises transforming the envelope representation residual coefficients into a warped domain so as to obtain transformed envelope representation residual coefficients. The method comprises applying, at least one of a plurality of gain-shape coding schemes on the transformed envelope representation residual coefficients in order to achieve gain-shape coded envelope representation residual coefficients, where the plurality of gain-shape coding schemes have mutually different trade-offs in one or more of gain resolution and shape resolution for one or more of the transformed envelope representation residual coefficients.

Abstract: Disclosed is a voice control system and method thereof. The voice control system is used in an electronic device and works in a sleep mode and a working mode. The voice control system comprises an audio detection module, an audio codec and a control module. Under the sleep mode, the audio detection module continually detects whether there is a wake-up speech in a received first audio data. If yes, the audio detection module generates a first indication signal and temporarily stores the following first audio data. When the control module is woken up by the first indication signal, the voice control system enters the working mode. Under the working mode, the control module drives the audio codec to read and process the temporarily stored first audio data to recognize control speech in the first audio data and to accordingly control the electronic device.

Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: A signal processing apparatus outputs a first coefficient stream that has values assigned to signal streams such that a maximum value of 1 or less is assigned to a signal stream closest to the first reference frequency among signal streams of harmonics appearing in frequency signal of sound, and for the other signal streams, the more distant from the first reference frequency the signal stream is, the smaller the value assigned to the signal stream becomes, generates a folding signal stream by multiplying the signal stream of the frequency signal having the bandwidth W on a lower frequency side than the first reference frequency by the first coefficient stream, and folding the signal stream of the frequency signal having a bandwidth W on the lower frequency side than the first reference frequency by using the first reference frequency as a symmetry axis, and outputs a complemented frequency signal.

Abstract: A computer-implemented method includes receiving, at a microphone of a voice-controlled device, a speech input from a user and determining, by the voice-controlled device, that a power state of an AV display device that is coupled to the voice-controlled device is an ON or OFF state. Based on the user intent determined from the speech input and the power state of the AV display, the voice-controlled device sends data to the AV display device to switch the AV display device ON or OFF. The method can further include receiving, by the voice-controlled device, content from a content source location and sending the content to the AV display device via an AV port.

Abstract: Provided are methods, systems, and apparatus for hierarchical decorrelation of multichannel audio. A hierarchical decorrelation algorithm is designed to adapt to possibly changing characteristics of an input signal, and also preserves the energy of the original signal. The algorithm is invertible in that the original signal can be retrieved if needed. Furthermore, the proposed algorithm decomposes the decorrelation process into multiple low-complexity steps. The contribution of these steps is generally in a decreasing order, and thus the complexity of the algorithm can be scaled.

Abstract: The invention provides methods and apparatuses for encoding a stereo audio signal having a left channel and a right channel. The apparatus includes a prediction coefficient estimator, downmixer, and multiplexer. The encoder is configured operate either in a prediction mode or a non-prediction mode.

Abstract: To suitably regulate sound pressure of object content on a receiving side. An audio stream including coded data of a predetermined number of pieces of object content is generated. A container of a predetermined format including the audio stream is transmitted. Information indicating a range within which sound pressure is allowed to increase and decrease for each piece of object content is inserted into a layer of the audio stream and/or a layer of the container. On a receiving side, sound pressure of each piece of object content increases and decreases within the allowable range based on the information.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

Abstract: A audio signal encoding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present disclosure, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

Abstract: An audio processing system includes a server complex in communication with a network. The server complex receives a digital audio file and one or more analog domain control settings from a client device across the network. A digital-to-analog converter converts the digital audio file to an analog signal. One or more analog signal processors apply at least one analog modification to the analog signal in accordance with the one or more analog domain control settings. An analog-to-digital converter converts the modified analog signal to a modified digital audio file. The server complex can then deliver the modified digital audio file to the client device across the network.

Abstract: The invention concerns a mobile terminal with at least two transducers (LSm, LSs1, LSS2) used simultaneously as loudspeakers for stereophonic effect. According to the invention, one of said transducers is a main transducer (LSm) with a main working frequency band (Bm) corresponding to at least the phone frequency band, while the other transducer is a secondary transducer (LSs1; LSs2) with a secondary working frequency band (Bs1; Bs2) band different from said main frequency band, the lowest frequencies of said secondary working frequency band (Bs1; Bs2) being greater than the lowest frequencies of said main working frequency band.

Abstract: This application relates to transfer of digital audio data between a host device (200) and an accessory apparatus (101) that may be connected to the host device via a suitable connector (106), such as a USB connector. A path selector (240, 270) is operable to establish either a first digital data path (201) or a second digital data path (202) for transfer of digital data. The first digital data path (201) includes a first data bus host (103) and a general purpose digital data interface (131) suitable for bulk data transfer between the first data bus host and the applications processor (110) of the device. This may be a default USB path. The second digital data path (202) includes a second data bus host (230) and at least one pair of second path data interfaces (140, 141). The second data bus host (230) does not form part of the applications processor and each of said second path data interfaces comprises a digital audio interface suitable for streaming of audio data.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: An apparatus for decomposing a signal having an number of at least three channels includes an analyzer for analyzing a similarity between two channels of an analysis signal related to the signal having at least two analysis channels, wherein the analyzer is configured for using a pre-calculated frequency dependent similarity curve as a reference curve to determine the analysis result. The signal processor processes the analysis signal or a signal derived from the analysis signal or a signal, from which the analysis signal is derived using the analysis result to obtain a decomposed signal.

Abstract: To suitably regulate sound pressure of object content on a receiving side. An audio stream including coded data of a predetermined number of pieces of object content is generated. A container of a predetermined format including the audio stream is transmitted. Information indicating a range within which sound pressure is allowed to increase and decrease for each piece of object content is inserted into a layer of the audio stream and/or a layer of the container. On a receiving side, sound pressure of each piece of object content increases and decreases within the allowable range based on the information.

Abstract: An example playback device may identify a first version of audio content for play back from a first source, and further identify a first value of an audio characteristic of the first version of the audio content, where the playback device includes a capability corresponding to the audio characteristic. The playback device may determine that the first value of the audio characteristic of the first version of the audio content does not match the capability of the playback device. The playback device may then identify a second version of the audio content from a second source, where the second version of the audio content includes the audio characteristic having a second value that is closer to the capability of the playback device. The playback device may then cause playback of the second version of the audio content.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (a) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: To address the technical problem, provided is a method of processing an audio signal, and the method including: receiving an audio bitstream that is encoded by using MPEG surround 212 (MPS212); obtaining internal channel gains with respect to one channel pair element (CPE) based on MPS212 parameters and rendering parameters about MPS212 output channels defined in a format converter; generating an internal channel signal with respect to the CPE based on the received audio bitstream and the obtained internal channel gains; and generating stereo output channel signals based on the generated internal channel signal.