A coding method, a decoding method, a coder, and a decoder, where the coding method includes obtaining the pulse distribution, on a track, of the pulses to be encoded on the track, determining a distribution identifier for identifying the pulse distribution according to the pulse distribution, and generating a coding index that includes the distribution identifier. The decoding method includes receiving a coding index, obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track, determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier, and reconstructing the pulse order on the track according to the pulse distribution.

The technology described in this document can be embodied in a computer-implemented method that includes receiving, at a first acoustic device, a representation of an audio signal, and amplifying the representation of the audio signal by a first gain factor to generate an amplified input signal. The method also includes processing the amplified input signal by an audio codec that includes one or more processors to generate a processed signal that represents a portion of the audio signal to be output by a second acoustic device. The processed signal includes noise originating at the audio codec. The method further includes transmitting the processed signal to the second acoustic device.

The technology described in this document can be embodied in a computer-implemented method that includes receiving, at a first acoustic device, a representation of an audio signal, and amplifying the representation of the audio signal by a first gain factor to generate an amplified input signal. The method also includes processing the amplified input signal by an audio codec that includes one or more processors to generate a processed signal that represents a portion of the audio signal to be output by a second acoustic device. The processed signal includes noise originating at the audio codec. The method further includes transmitting the processed signal to the second acoustic device.

An encoder and a method therein for Pyramid Vector Quantizer, PVQ, shape search, the PVQ taking a target vector x as input and deriving a vector y by iteratively adding unit pulses in an inner dimension search loop. The method comprises, before entering a next inner dimension search loop for unit pulse addition, determining, based on the maximum pulse amplitude, maxamp.sub.y, of a current vector y, whether more than a current bit word length is needed to represent enloop.sub.y, in a lossless manner in the upcoming inner dimension loop. The variable enloop.sub.y is related to an accumulated energy of the vector y. The performing of this method enables the encoder to keep the complexity of the search at a reasonable level.

The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

In the field of audio digital processing, an Android-based audio content processing method and device are provided. In an embodiment, the method includes: receiving, by a framework layer, audio content; identifying, by the framework layer, a type of the audio content, and adding an identifier associated with the type to the identified audio content; and receiving, by a hardware abstraction layer (HAL), audio content data and the identifier from the framework layer, and sending the audio content data to a processing unit corresponding to the identifier. In the embodiment, an Android framework layer is used to identify a type of audio content, and mark the type, and an HAL then sends, according to a mark, pulse coding modulation (PCM) data corresponding to the audio content to a processing unit corresponding to the type of the audio content, thereby implementing correct processing on audio content of different types. By means of the foregoing technical solution, coexistence of Dolby, a digital theater system (DTS), and high-fidelity (HiFi) in a same system is implemented.

An encoding method and encoder is provided for transparent lossless audio watermarking by quantising an original PCM audio signal twice, each quantisation quantising to a quantisation grid. As a PCM signal is inherently already quantised, there are three quantisation grids to consider, the first being the quantisation grid of the original PCM signal, the second being that of the watermarked signal and the third being that of an intermediate signal. The technique reduces the amount of introduced quantisation error, spectrally shapes the error and fully decorrelates signal alterations from the original audio, thus making the error more similar to additive noise. A decoding method and decoder is also provided, as is a method of altering the watermark without fully decoding the encoded signal.

An encoding method and encoder is provided for transparent lossless audio watermarking by quantising an original PCM audio signal twice, each quantisation quantising to a quantisation grid. As a PCM signal is inherently already quantised, there are three quantisation grids to consider, the first being the quantisation grid of the original PCM signal, the second being that of the watermarked signal and the third being that of an intermediate signal. The technique reduces the amount of introduced quantisation error, spectrally shapes the error and fully decorrelates signal alterations from the original audio, thus making the error more similar to additive noise. A decoding method and decoder is also provided, as is a method of altering the watermark without fully decoding the encoded signal.

The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.