Samsung Electronics Co., Ltd.

Motorola Mobility LLC

Google Inc.

Motorola Solutions, Inc.

Holly Francois

Eun-mi Oh

Jonathan A Gibbs

Ki-hyun Choo

Kyung-hun Jung

Ho-sang Sung

Ho-Sang Sung

Yaodu Wei

James Patrick Ashley

Udar Mittal

Robert A Zurek

Gregory Joseph Dunn

Joel A Clark

Sangjun Park

Hosang Sung

Chang-Yeong Kim

Kihyun Choo

Seungdo Choi

Terance B Blake

Peter Lin

James Alexander Rex

Scott K Isabelle

David John Pearce

Jeffrey M Axelrod

Jesus Monge Alvarez

A system for synthesising expressive speech includes: an interface configured to receive an input text for conversion to speech; a memory; and at least one processor coupled to the memory. The processor is configured to generate, using an expressivity characterisation module, a plurality of expression vectors, wherein each expression vector is a representation of prosodic information in a reference audio style file, and synthesise expressive speech from the input text, using an expressive acoustic model comprising a deep convolutional neural network that is conditioned by at least one of the plurality of expression vectors.

Expressive text-to-speech system and method

Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Signal processing methods and apparatuses for enhancing sound quality

Provided is a method for processing a signal adaptively to a noisy environment, the method including: determining a class related to enhancement of intelligibility, based on a noise spectrum of a reception side or a near-end device and a received voice spectrum; generating guide information for the enhancement of intelligibility based on the noise spectrum, the voice spectrum, and the determined class; and generating a changed voice spectrum by applying the guide information to the voice spectrum.

Signal processing method and device adaptive to noise environment and terminal device employing same

A voice signal processing method according to an embodiment of the present disclosure for overcoming the problem includes: acquiring a real-time near-end noise signal; acquiring a far-end voice signal according to an incoming call; measuring subjective speech quality and perceptual-objective speech quality of test signals generated based on a reference signal and the real-time near-end noise signal; selecting at least one speech quality enhancement method based on the subjective speech quality and the perceptual-objective speech quality, and determining parameters that are to be applied to the selected at least one speech quality enhancement method; and enhancing speech quality of the far-end voice signal by using the selected at least one speech quality enhancement method, based on the determined parameters, wherein the test signals are generated by mixing the acquired real-time near-end noise signal with the reference signal whose speech quality is enhanced by applying a combination of parameter values to speech quality enhancement methods.

Method and apparatus for improving call quality in noise environment

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

Apparatus () for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor () which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k&#x2212;1 subsequent frames to values, such that the values of the n bits of the k&#x2212;1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Apparatus and method for encoding at least one parameter associated with a signal source

A method and apparatus provides for encoding an audio signal. A bit rate value is received. A set of energy thresholds based on the bit rate value is selected. The set of energy thresholds is one of a plurality of sets of energy thresholds. The energy thresholds of each set of energy thresholds correspond on a one-to-one basis with a set of sub-bands of the audio signal. The audio signal is received. The energy of each sub-band of the set of sub-bands is determined. A highest frequency sub-band that has an energy exceeding the corresponding threshold is determined. A selected bandwidth of the audio signal is encoded. The selected bandwidth includes only those frequencies of the audio signal that are in the highest frequency sub-band that has an energy exceeding the corresponding threshold, as well as the lower frequencies of the audio signal that are above a high-pass cut-off frequency.

Apparatus and method for audio encoding

A method for estimating a parameter for low bit rate stereo transmission that includes deriving estimate of any time delay between left and right audio channels in a multi-channel signal from a time delay subsystem. A cross-correlation between the left and right audio channels in the time delay subsystem is employed. Thereafter a normalized cross-correlation within an inter-channel intensity difference (IID) processor is employed before deriving estimate of panning gains for the left and right audio channels from the IID processor.

Method and apparatus for estimating a parameter for low bit rate stereo transmission

A system, method, and apparatus for separating speech signal from a noisy acoustic environment. The separation process may include directional filtering, blind source separation, and dual input spectral subtraction noise suppressor. The input channels may include two omnidirectional microphones whose output is processed using phase delay filtering to form speech and noise beamforms. Further, the beamforms may be frequency corrected. The omnidirectional microphones generate one channel that is substantially only noise, and another channel that is a combination of noise and speech. A blind source separation algorithm augments the directional separation through statistical techniques. The noise signal and speech signal are then used to set process characteristics at a dual input noise spectral subtraction suppressor (DINS) to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify speech signal.

Robust two microphone noise suppression system

A communication device contains multiple microphones that receive acoustic signals from a user and from the background. The acoustic signals from the user are enhanced using the background acoustic signals to reduce background noise. The enhanced signal are transmitted to an emergency network when an emergency call is made from the communication device. The raw signals are stored in the communication device for later retrieval or are transmitted simultaneously with the enhanced signals. The enhanced signals are transmitted using a circuit-switched voice mode while the raw signals are transmitted using a packet-switched voice mode.

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Guildford, United Kingdom

Holly Francois