A data encoding system includes a non-transitory memory, a processor, a digital-to-analog converter (DAC) and a transmitter. The non-transitory memory stores a predetermined file size threshold. The processor is in operable communication with the memory, and is configured to receive data. The processor detects a file size associated with the data. When the file size is below the predetermined file size threshold, the processor compresses the data using a variable length codeword (VLC) encoder. When the file size is not below the predetermined file size threshold, the processor compresses the data, using a hash table algorithm. The DAC is configured to receive a digital representation of the compressed data from the processor and convert the digital representation of the compressed data into an analog representation of the compressed data. The transmitter is coupled to the DAC and configured to transmit the analog representation of the compressed data.

An encoder includes a low-pass filter to filter input audio signals. The low-pass filter has fixed filter coefficients. The encoder generates quantized signals based on a difference signal. The encoder includes an adaptive quantizer and a decoder to generate feedback signals. The decoder has an inverse quantizer and a predictor. The predictor has fixed control parameters which are based on a frequency response of the low-pass filter. The predictor may include a finite impulse response filter having fixed filter coefficients. The decoder may include an adaptive noise shaping filter coupled between the low-pass filter and the encoder. The adaptive noise shaping filter flattens signals within a frequency spectrum corresponding to a frequency spectrum of the low-pass filter.

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.

To enable to favorably send a compressed digital audio signal at a high data rate. First, second, and third metadata are added to a compressed digital audio signal of a predetermined number of channels. The first metadata is metadata indicating a sending frequency of the compressed digital audio signal. The second metadata is metadata indicating a sampling frequency used for converting an uncompressed digital audio signal of each channel into an analog signal. The third metadata is metadata indicating a ratio of the sending frequency to the sampling frequency. The compressed digital audio signal provided with each type of the metadata is transmitted to an external device through a predetermined sending path.

A source device comprising a memory and a processor may be configured to perform techniques described in this disclosure. The memory may store at least a portion of the audio data. The processor may obtain, from a compressed version of the audio data, a symbol, and obtain a plurality of intervals, each having a same bit length. The processor may obtain a portion of the symbol within the bit length and an excess portion of the symbol over the bit length, and specify, in a first interval, the portion of the symbol. The processor may also specify, in a second interval, the excess portion of the symbol, and apply, to the first interval and the second interval, error resiliency. The processor may specify, in a bitstream representative of the compressed version of the audio data, the first error resilient interval and the second error resilient interval.

A data encoding system includes a non-transitory memory, a processor, a digital-to-analog converter (DAC) and a transmitter. The non-transitory memory stores a predetermined file size threshold. The processor is in operable communication with the memory, and is configured to receive data. The processor detects a file size associated with the data. When the file size is below the predetermined file size threshold, the processor compresses the data using a variable length codeword (VLC) encoder. When the file size is not below the predetermined file size threshold, the processor compresses the data, using a hash table algorithm. The DAC is configured to receive a digital representation of the compressed data from the processor and convert the digital representation of the compressed data into an analog representation of the compressed data. The transmitter is coupled to the DAC and configured to transmit the analog representation of the compressed data.

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.

To enable to favorably send a compressed digital audio signal at a high data rate. First, second, and third metadata are added to a compressed digital audio signal of a predetermined number of channels. The first metadata is metadata indicating a sending frequency of the compressed digital audio signal. The second metadata is metadata indicating a sampling frequency used for converting an uncompressed digital audio signal of each channel into an analog signal. The third metadata is metadata indicating a ratio of the sending frequency to the sampling frequency. The compressed digital audio signal provided with each type of the metadata is transmitted to an external device through a predetermined sending path.

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.