Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.

Spectral encoding methods are more robust when used with improved weak signal detection and synchronizations methods. Further robustness gains are achieved by using informed embedding, error correction and embedding protocols that enable signal to noise enhancements by folding and pre-filtering the received signal.

A speech recognition method is provided that recognizes speech for causing equipment to operate. The method includes acquiring a speech signal from a microphone disposed in a designated space. The method also includes detecting a spatial sound pressure distribution indicating a distribution of sound pressure in the space, on the basis of the acquired speech signal, and detecting a point sound source in the space on the basis of the detected spatial sound pressure distribution. The method further includes judging to conduct a speech recognition process on the acquired speech signal when the point sound source is detected.

Described is an apparatus which comprises: first logic to generate a first audio data and to embed the first audio data with a watermark to generate an embedded data; a speaker to output the embedded data as a first audible audio; a microphone to receive a second audible audio; and second logic to check the second audible audio for the watermark, and if the second audible audio has the watermark embedded in the first audio data, generate a first message, else generate a second message.

Techniques are disclosed for adjusting user experience of a software application based on paralinguistic information. One embodiment presented herein includes a computer-implemented method for adjusting a user experience of a software application. The method comprises receiving, at a computing device, an audio stream comprising audio of a user. The method further comprises analyzing the audio stream for paralinguistic information to determine an attribute of the user. The method further comprises identifying content of the audio stream. The method further comprises determining one or more actions based on the content of the audio stream. The method further comprises selecting at least one of the one or more actions based on the attribute of the user.

A method (500) for resampling an audio signal (110) is described. The method (500) comprising providing (501) a set of input subband signals (210) which is representative of a time domain audio signal. Furthermore, the method (500) comprises applying (502) a first ripple pre-emphasis gain (323) to a first input subband signal (210) of the set of input subband signals (210) to determine a corresponding first output subband signal (213) of a set of output subband signals (213). In addition, the method (500) comprises determining (503) a time domain input audio signal (110) from the set of output subband signals (213). The method (500) further comprises performing (504) time domain resampling of the input audio signal (110) to provide an output audio signal (113) using an anti-aliasing filter (102), wherein the first ripple pre-emphasis gain (323) is dependent on a frequency response (311) of the anti-aliasing filter (102), such that an amplitude ripple of the frequency response (311) of the anti-aliasing filter (102) is at least partially compensated by the first ripple pre-emphasis gain (323).

A Mobile Voice Self Service (MVSS) mobile device and method thereof. A VoiceXML browser that is implemented directly on the MVSS mobile device may request a VoiceXML application and process it. A call data manager may also be implemented on the MVSS mobile device and may provide call data that may authorize access to advanced Media Resource Control Protocol (MRCP) services, such as Automatic Speech Recognition (ASR) or Text-To-Speech (TTS). A media resource gateway may then provide the advanced MRCP services to the VoiceXML application processed by the VoiceXML application browser. Hotkey navigations and bookmarked application points to VoiceXML applications may be created and applied through application analysis and state tracking. Therein, VoiceXML document transitions and user input are stored to maintain application state changes until the user requests creation of an application bookmark.

A Mobile Voice Self Service (MVSS) mobile device and method thereof. A VoiceXML browser that is implemented directly on the MVSS mobile device may request a VoiceXML application and process it. A call data manager may also be implemented on the MVSS mobile device and may provide call data that may authorize access to advanced Media Resource Control Protocol (MRCP) services, such as Automatic Speech Recognition (ASR) or Text-To-Speech (TTS). A media resource gateway may then provide the advanced MRCP services to the VoiceXML application processed by the VoiceXML application browser. Hotkey navigations and bookmarked application points to VoiceXML applications may be created and applied through application analysis and state tracking. Therein, VoiceXML document transitions and user input are stored to maintain application state changes until the user requests creation of an application bookmark.

Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are provided. A plurality of audio signals that are continuous is obtained. It is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type.

An audio processing system (100) comprises a front-end component (102, 103), which receives quantized spectral components and performs an inverse quantization, yielding a time-domain representation of an intermediate signal. The audio processing system further comprises a frequency-domain processing stage (104, 105, 106, 107, 108), configured to provide a time-domain representation of a processed audio signal, and a sample rate converter (109), providing a reconstructed audio signal sampled at a target sampling frequency. The respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the time-domain representation of the processed audio signal are equal. In particular embodiments, the processing stage comprises a parametric upmix stage which is operable in at least two different modes and is associated with a delay stage that ensures constant total delay.