Systems and methods for packet payload mapping for robust transmission of data are described. For example, methods may include receiving, using a network interface, packets that each respectively include a primary frame and one or more preceding frames from the sequence of frames of data that are separated from the primary frame in the sequence of frames by a respective multiple of a stride parameter; storing the frames of the packets in a buffer with entries that each hold the primary frame and the one or more preceding frames of a packet; reading a first frame from the buffer as the primary frame from one of the entries; determining that a packet with a primary frame that is a next frame in the sequence has been lost; and, responsive to the determination, reading the next frame from the buffer as a preceding frame from one of the entries.

This application relates to methods and apparatus for transfer of multiple digital data streams, especially of digital audio data over a single communications link such as a single wire. The application describes audio interface circuitry comprising a pulse-length-modulation (PLM) modulator. The PLM is responsive to a plurality of data streams (PDM-R, PDM-L), to generate a series of data pulses (PLM) with a single data pulse having a rising and falling edge in each of a plurality of transfer periods defined by a first clock signal (TCLK). The timing of the rising and falling edge of each data pulse is dependent upon a combination of the then current data samples from the plurality of data streams. The duration and position of the data pulse in the transfer window in effect defines a data symbol encoding the data. Circuitry for receiving and extracting the data is also disclosed. An interface receives the stream of data pulses (PLM) and data extraction circuitry samples the data pulse to determine which of the possible data symbols the pulse represents and determines a data value for at least one received data stream.

An audio encoding apparatus comprises an audio receiver (201) receiving audio items representing an audio scene and a metadata receiver (203) receives input presentation metadata for the audio items describing presentation constraints for the rendering of the audio items. The presentation constraints constrain a rendering parameter that can be adapted when rendering the audio items. An audio encoder (205) generates encoded audio data for the audio scene by encoding the plurality of audio items with the encoding being adapted in response to the input presentation metadata. A metadata circuit (207) generates output presentation metadata from the input presentation metadata. The output presentation metadata comprises data for encoded audio items which constrain the extent by which an adaptable parameter of a rendering can be adapted when rendering the encoded audio items. An output (209) generates an encoded audio data stream comprising the encoded audio data and the output presentation metadata.

A system and method to process audio data received over the ARC or eARC interface of HDMI from audio sources are provided. A media device may receive compressed audio data in a number of data formats. The media device may convert between the audio formats provided by the audio sources and the audio formats supported by audio playback devices. The media device may inspect frames of audio data to determine if the frames are to be decoded. The frame may be decoded and subsequently encoded into the data formats supported by the audio playback devices. To reduce latency, the media device may enable a pass-through mode to bypass the decoding of the frames to allow the frames to be decoded at the audio playback devices. A bi-directional loopback application may route audio data received over the ARC or eARC interface from the audio sources to the audio playback devices.

Disclosed are methods and systems for using softbit decoding techniques in retransmission-based networks for error concealment of packets corrupted by bit-errors. The softbit decoding techniques derive softbit information from multiple corrupted hardbits of the retransmitted packet to aid a softbit decoder in decoding the packet. The approach realizes improved error concealment capability while maintaining a simple system architecture. A retransmission softbit module is inserted between a channel decoder used for channel-decoding and demodulating a compressed packet and the softbit decoder. The retransmission softbit module may derive an accumulated softbit packet from multiple corrupted copies of the packet received from the channel decoder, make bit decisions based on the accumulated softbit packet, and derive reliability information for the bit decisions. The bit decisions may be a majority decision packet (MDP) created using a majority voting scheme. The reliability information and the MDP may be provided to the softbit decoder for decoding.

Disclosed are methods of encoding and decoding an audio signal, and an encoder and a decoder for performing the methods. The method of encoding an audio signal includes identifying an input signal corresponding to a low frequency band of the audio signal, windowing the input signal, generating a first latent vector by inputting the windowed input signal to a first encoding model, transforming the windowed input signal into a frequency domain, generating a second latent vector by inputting the transformed input signal to a second encoding model, generating a final latent vector by combining the first latent vector and the second latent vector, and generating a bitstream corresponding to the final latent vector.

It is possible to enable a reception side to easily recognize that metadata is inserted into an audio stream. A metafile including meta information for acquiring an audio stream into which metadata is inserted through a reception device is transmitted. The identification information indicating that the metadata is inserted into the audio stream is inserted into the metafile. At the reception side, it is possible to easily recognize that the metadata is inserted into the audio stream based on the identification information inserted into the metafile.

The present technology relates to an encoding device and method, a decoding device and method, and a program, which are adapted to be capable of improving convenience. The decoding device is provided with: a decoding unit that decodes audio data including an object audio, the audio data being included in an encoded bit stream, and reads metadata of the object audio from an area in which arbitrary data of the encoded bit stream can be stored; and an output unit that outputs the decoded audio data on the basis of the metadata. The present technology can be applied to the decoding device.

Aspects of the disclosure include methods, apparatuses, and non-transitory computer-readable storage mediums for loudness adjustment for an audio scene associated with an MPEG-I immersive audio stream. One apparatus includes processing circuitry that receives a first syntax element indicating a number of sound signals included in the audio scene. The processing circuitry determines whether one or more speech signals are included in the sound signals indicated by the first syntax element. The processing circuitry determines a reference speech signal from the one or more speech signals based on the one or more speech signals being included in the sound signals. The processing circuitry adjusts a loudness level of the reference speech signal of the audio scene based on an anchor speech signal. The processing circuitry adjusts loudness levels of the sound signals based on the adjusted loudness level of the reference speech signal.

Audio streaming devices, systems, and methods may employ adaptive differential pulse code modulation (ADPCM) techniques providing for optimum performance even while ensuring robustness against transmission errors. One illustrative device includes: a difference element that produces a sequence of prediction error values by subtracting predicted values from audio samples; a scaling element that produces scaled error values by dividing each prediction error by a corresponding envelope estimate; a quantizer that operates on the scaled error values to produce quantized error values; a multiplier that uses the corresponding envelope estimates to produce reconstructed error values; a predictor that produces the next audio sample values based on the reconstructed error values; and an envelope estimator. The envelope estimator includes: an updater that applies a dynamic gain to the reconstructed error values to produce update values; and an integrator that combines each of the update values with the corresponding envelope estimate to produce a subsequent envelope estimate.