A system and method are presented for the correction of packet loss in audio in automatic speech recognition (ASR) systems. Packet loss correction, as presented herein, occurs at the recognition stage without modifying any of the acoustic models generated during training. The behavior of the ASR engine in the absence of packet loss is thus not altered. To accomplish this, the actual input signal may be rectified, the recognition scores may be normalized to account for signal errors, and a best-estimate method using information from previous frames and acoustic models may be used to replace the noisy signal.

A system and method are presented for the correction of packet loss in audio in automatic speech recognition (ASR) systems. Packet loss correction, as presented herein, occurs at the recognition stage without modifying any of the acoustic models generated during training. The behavior of the ASR engine in the absence of packet loss is thus not altered. To accomplish this, the actual input signal may be rectified, the recognition scores may be normalized to account for signal errors, and a best-estimate method using information from previous frames and acoustic models may be used to replace the noisy signal.

A coding method and a decoding method are provided which can use in combination a predictive coding and decoding method which is a coding and decoding method that can accurately express coefficients which are convertible into linear prediction coefficients with a small code amount and a coding and decoding method that can obtain correctly, by decoding, coefficients which are convertible into linear prediction coefficients of the present frame if a linear prediction coefficient code of the present frame is correctly input to a decoding device. A coding device includes: a predictive coding unit that obtains a first code by coding a differential vector formed of differentials between a vector of coefficients which are convertible into linear prediction coefficients of more than one order of the present frame and a prediction vector containing at least a predicted vector from a past frame, and obtains a quantization differential vector corresponding to the first code; and a non-predictive coding unit that generates a second code by coding a correction vector which is formed of differentials between the vector of the coefficients which are convertible into the linear prediction coefficients of more than one order of the present frame and the quantization differential vector or formed of some of elements of the differentials.

In a reliable multi-cast, a concealment scheme may be applied to recover or conceal lost or otherwise corrupted packets of audio information for one channel based on the audio information of other channels in the reliable multi-cast. The concealment scheme may employ correction factors for channels derived from the channel relationships.

In a reliable multi-cast, a concealment scheme may be applied to recover or conceal lost or otherwise corrupted packets of audio information for one channel based on the audio information of other channels in the reliable multi-cast. The concealment scheme may employ correction factors for channels derived from the channel relationships.

The present disclosure provides a method, an apparatus and a device for processing a sound signal, wherein the method comprises: acquiring a transmitted signal spectrum of a target sound signal sent out by a loudspeaker and a received signal spectrum of the target sound signal received by a microphone; detecting whether there is a signal distortion frequency band with signal distortion in the target sound signal according to the transmitted signal spectrum and the received signal spectrum, and when detecting that the signal distortion frequency band exists, performing compression processing on the target sound signal according to the signal distortion frequency band during a current signal processing cycle, and transmitting a compressed target sound signal through the loudspeaker.

The present disclosure provides a method, an apparatus and a device for processing a sound signal, wherein the method comprises: acquiring a transmitted signal spectrum of a target sound signal sent out by a loudspeaker and a received signal spectrum of the target sound signal received by a microphone; detecting whether there is a signal distortion frequency band with signal distortion in the target sound signal according to the transmitted signal spectrum and the received signal spectrum, and when detecting that the signal distortion frequency band exists, performing compression processing on the target sound signal according to the signal distortion frequency band during a current signal processing cycle, and transmitting a compressed target sound signal through the loudspeaker.

Methods and systems for improving the robustness of wireless communications. The methods and systems provided transmit data packets over a first isochronous stream and transmit one or more supplemental data packets over the same time intervals. The one or more supplemental data packets are used to re-create and/or enhance at least a portion of one or more data packets of the plurality of data packets that have already been sent. Alternatively, the one or more supplemental data packets are used to create and/or enhance at least a portion of one or more data packets of the plurality of data packets that will be received during the next isochronous intervals. The methods and system described herein allow for increased robustness by allowing for better retransmission with correctly received packets and the methods set forth herein work with any Bluetooth broadcaster sink without modification.

Methods and systems for improving the robustness of wireless communications. The methods and systems provided transmit data packets over a first isochronous stream and transmit one or more supplemental data packets over the same time intervals. The one or more supplemental data packets are used to re-create and/or enhance at least a portion of one or more data packets of the plurality of data packets that have already been sent. Alternatively, the one or more supplemental data packets are used to create and/or enhance at least a portion of one or more data packets of the plurality of data packets that will be received during the next isochronous intervals. The methods and system described herein allow for increased robustness by allowing for better retransmission with correctly received packets and the methods set forth herein work with any Bluetooth broadcaster sink without modification.

An audio codec suitable for robust wireless transmission of high quality audio with low latency, still at a moderate bit rate. The encoding and decoding methods are based on ADPCM and in addition to the encoded output bits APM, additional data QB are included in output data blocks, namely data QB representing an internal value of the adaptive quantization ADQ of the ADPCM encoding algorithm, especially a scaling factor encoded and truncated to such as 8 bits. Further, output data blocks preferably include data CFB representing an internal value of the predictor PR of the ADPCM encoding algorithm, especially data CFB representing coefficients of a lattice prediction FIR filter which, truncated to such as 8 bits, can be sequentially included in output data blocks. These additional data QB, CFB regarding internal values of the ADPCM encoding algorithm can be utilized at the encoder side to increase robustness against loss of data blocks in wireless transmission. Especially, the decoding algorithm may comprise comparing its current internal ADPCM decoding values corresponding to the received internal values QB, CFB from the encoder, and in case there is a difference, the decoder can adapt or overwrite its internal values to the ones received QB, CFB. This helps to ensure fast recovery after lost data blocks, thereby ensuring robustness against artefacts in the reconstructed signal, e.g. clicks in case of audio.