An audio decoder for providing a decoded audio information includes a arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder is configured to select a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state. The arithmetic decoder is configured to determine or modify the current context state in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder is configured to detect a group of a plurality of previously-decoded spectral values, which fulfill, individually or taken together, a predetermined condition regarding their magnitudes, and to determine the current context state in dependence on a result of the detection.

An audio encoder uses similar principles.

A method for loudness equalization is provided that includes receiving input loudness data at an audio processing system. Converting gain data of the input loudness data to a linear scale at the audio processing system. Determining a reciprocal of a gain-linear loudness value as a function of the converted gain data using the audio processing system. Determining a compression ratio using the audio processing system. Performing temporal smoothing and look ahead processing using the audio processing system. Outputting gain data as a function of the temporal smoothing and look ahead processing using the audio processing system.

A system and method for provide high quality audio in real-time communication over low bit rate network connections. The system includes real-time communication software application having an improved encoder and an improved decoder. The encoder decomposes audio data based on two frequency ranges corresponding to a super wideband mode and a wideband mode into a lower sub-band and a higher sub-band. Audio features are extracted from the lower sub-band and higher sub-band audio data. The audio features are quantized and packaged. The decoder reconstructs the audio data for playback on the receiving device based on the compressed audio features in the super wideband mode and the wideband mode.

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. The method comprises determining a model parameter (613) of a signal model; determining a prediction coefficient to be applied to a previous sample (614) of a first decoded subband signals derived from the first subband signal, based on the signal model, based on the model parameter (613) and based on the analysis filterbank (612); wherein a time slot of the previous sample (614) is prior to a time slot of the first sample (615); and determining an estimate of the first sample (615) by applying the prediction coefficient to the previous sample (614).

The present disclosure relates to processing a plurality of audio signals. A device receives the plurality of audio signals in the frequency domain and determining an overall attenuation multiplier based on the plurality of audio signals and an overall lookup table that relates decibel values to different overall attenuation multipliers. The device determines an attenuation vector comprising a plurality of bin-specific attenuation multipliers, each bin-specific attenuation multiplier respectively corresponding to a different frequency bin of the plurality of frequency bins. The device scales each bin-specific attenuation value in the attenuation vector with the overall attenuation multiplier, and edits each of the audio signals based on the scaled bin-specific attenuation values in the attenuation vector.

The present document relates to audio encoding and decoding. In particular, the present document relates to audio coding schemes which make use of high frequency reconstruction (HFR) methods. A system configured to determine a master scale factor band table of a highband signal (105) of an audio signal is described. The highband signal (105) is to be generated from a lowband signal (101) of the audio signal using a high frequency reconstruction (HFR) scheme. The master scale factor band table is indicative of a frequency resolution of a spectral envelope of the highband signal (105).

A system, apparatus and a method for transmitting continuous audio data configured to mitigate data discontinuities in a receiving device. The method may mitigate data discontinuities by transmitting a continuous stream of audio data that has reduced changes to the audio data characteristics. The method may transmit filler audio data when no application audio data is available. The application audio data and the filler audio data are processed to reduce changes to the audio data characteristics in each stream.

In some examples, an audio sending device receives a stream of application audio data, encodes the stream of application audio data, and in response to detecting an end of the stream of application audio data, provides pre-encoded filler audio data from a buffer in the audio sending device as an encoded stream of filler audio data. The audio sending device transmits the encoded stream of application audio data and the encoded stream of filler audio data in an encoded output data stream over a transport to an audio receiving device.

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.

Example methods and apparatus to audio watermarking and watermark detection and extraction are disclosed herein. An example apparatus disclosed herein includes memory, computer readable instructions, and processor circuitry to execute the computer readable instructions to at least detect a first symbol, a second symbol, a third symbol, and a fourth symbol sequentially in encoded audio samples, determine whether the first symbol is a synchronization symbol, in response to a determination that the first symbol is a synchronization symbol, determine that the first symbol and the third symbol are associated with a first message and the second symbol and the fourth symbol are associated with a second message, and output at least one of the first message or the second message.