Motorola Mobility LLC

Motorola Corporation

Huawei Technologies Co., Limited

Google Inc.

Google Technology Holdings LLC

Jonathan A Gibbs

Bin Wang

James Patrick Ashley

Udar Mittal

Eyal Shlomot

Haiting Li

Holly Francois

Richard C Lucas

Nicholas William Whinnett

William Neil Robinson

Halting Li

Anthony P Van Den Heuvel

Andrew John Aftelak

James P Ashley

Mark A Jasiuk

Michael Joseph McLaughlin

David Nigel Freeman

Shaun Richard Fenton

David John Pearce

William Robinson

Antony John Wray

Dragan Boscovic

Halil Fikretler

Tiong Tan Boon

Christopher Philip Johnson

Timor Kadir

Anthony Van Den Heuvel

Dominic Chan

Holly L Francois

A stereo signal encoding method includes performing spectrum broadening on a quantized line spectral frequency (LSF) parameter of a primary channel signal in a current frame in a stereo signal to obtain a spectrum-broadened LSF parameter of the primary channel signal, determining a prediction residual of an LSF parameter of a secondary channel signal in the current frame based on an original LSF parameter of the secondary channel signal and the spectrum-broadened LSF parameter of the primary channel signal, and performing a quantization on the prediction residual of the LSF parameter of the secondary channel signal.

Stereo signal encoding method and apparatus, and stereo signal decoding method and apparatus

An encoding method includes determining an adaptive broadening factor based on a quantized line spectral frequency (LSF) vector of a first channel of a current frame of an audio signal and an LSF vector of a second channel of the current frame, and writing the quantized LSF vector and the adaptive broadening factor into a bitstream.

Audio signal encoding method and apparatus

In an audio signal processing method, a processing device obtains a current position relationship between a sound source and a listener. The processing device then obtains a current audio rendering function based on the current position relationship. When the current position relationship is different from a stored previous position relationship, the processing device adjusts an initial gain of the current audio rendering function based on the current position relationship and the previous position relationship, to obtain an adjusted gain of the current audio rendering function. The processing device then obtains an adjusted audio rendering function based on the current audio rendering function and the adjusted gain, and generates a current output signal based on a current input signal and the adjusted audio rendering function.

Audio signal processing method and apparatus

A stereo signal encoding method includes determining a window length of an attenuation window based on an inter-channel time difference; determining a modified linear prediction analysis window based on the window length of the attenuation window, where values of at least some points from a point (L&#x2212;sub_window_len) to a point (L&#x2212;1) in the modified linear prediction analysis window are less than values of corresponding points from a point (L&#x2212;sub_window_len) to a point (L&#x2212;1) in an initial linear prediction analysis window, and the window length of the modified linear prediction analysis window is equal to a window length of the initial linear prediction analysis window; and performing linear prediction analysis on a to-be-processed sound channel signal based on the modified linear prediction analysis window.

Stereo signal encoding method and encoding apparatus

An encoding method includes determining a target adaptive broadening factor based on a quantized line spectral frequency (LSF) parameter of a primary channel signal in a current frame and an LSF parameter of a secondary channel signal in the current frame, and writing the quantized LSF parameter of the primary channel signal in the current frame and the target adaptive broadening factor into a bitstream.

A stereo signal encoding method includes determining a window length of an attenuation window based on an inter-channel time difference, determining a modified linear prediction analysis window based on the window length of the attenuation window, where values of at least some points from a point (L&#x2212;sub_window_len) to a point (L&#x2212;1) in the modified linear prediction analysis window are less than values of corresponding points from a point (L&#x2212;sub_window_len) to a point (L&#x2212;1) in an initial linear prediction analysis window, and the window length of the modified linear prediction analysis window is equal to a window length of the initial linear prediction analysis window, and performing linear prediction analysis on a to-be-processed sound channel signal based on the modified linear prediction analysis window.

An encoding apparatus comprises a frame processor () which receives a multi channel audio signal comprising at least a first audio signal from a first microphone () and a second audio signal from a second microphone (). An ITD processorthen determines an inter time difference between the first audio signal and the second audio signal and a set of delays () generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner () then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder () encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Apparatus and method for encoding a multi-channel audio signal

An apparatus receives an input encoded audio data signal comprising a base layer and at least one enhancement layer. A reference unit () generates reference audio data corresponding to audio data of a reference set of layers. A layer unit () divides the layers of the input signal into a first subset and a second subset. A sample unit () generates sample audio data corresponding to the audio data of the first subset. A comparison unit () generates a difference measure by comparing the sample audio data to the reference audio data based on a perceptual model. An output unit () then determines if the difference measure meets a similarity criterion and generates an output signal without audio data from a layer of the second subset if the similarity criterion is met and including the audio data of the layer otherwise. The invention may provide reduced data rates without an unacceptable degradation of quality.

Apparatus and method for generating an output audio data signal

A hybrid speech encoder detects changes from music-like sounds to speech-like sounds. When the encoder detects music-like sounds (e.g., music), it operates in a first mode, in which it employs a frequency domain coder. When the encoder detects speech-like sounds (e.g., human speech), it operates in a second mode, and employs a time domain or waveform coder. When a switch occurs, the encoder backfills a gap in the signal with a portion of the signal occurring after the gap.

Method and apparatus for encoding an audio signal

A method and apparatus for encoding a signal is provided herein. During operation a wideband signal that is to be encoded enters a filter bank. A highband signal and a lowband signal are output from the filter bank. Each signal is separately encoded. During the production of the highband signal, a downmixing operation is implemented after preprocessing, and prior to decimating. The downmixing operation greatly reduces system complexity. In fact, it will be observed that the highest sample rate in the prior-art implementation is 64 kHz whereas the sample rate in the system described above remains at 32 kHz or below. This represents a significant complexity saving, as do the reduced number of processing blocks.

Method and apparatus for encoding a wideband speech signal utilizing downmixing of a highband component

An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames.

Method and apparatus for audio coding and decoding

A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Growing community of inventors

Winchester, United Kingdom

Jonathan A Gibbs