Audio streaming devices, systems, and methods may employ adaptive differential pulse code modulation (ADPCM) techniques providing for optimum performance even while ensuring robustness against transmission errors. One illustrative device includes: a difference element that produces a sequence of prediction error values by subtracting predicted values from audio samples; a scaling element that produces scaled error values by dividing each prediction error by a corresponding envelope estimate; a quantizer that operates on the scaled error values to produce quantized error values; a multiplier that uses the corresponding envelope estimates to produce reconstructed error values; a predictor that produces the next audio sample values based on the reconstructed error values; and an envelope estimator. The envelope estimator includes: an updater that applies a dynamic gain to the reconstructed error values to produce update values; and an integrator that combines each of the update values with the corresponding envelope estimate to produce a subsequent envelope estimate.

A method of using stereo recording to control the flashing of central lamps includes the steps of: providing left and right channel microphones in a stereo sound source venue or electronic device, and recording the plural sound sources generated by the left area, central area and right area of the stereo sound source, then storing the recorded analog signal in an audio signal sampling unit of a signal processing device, and then converting the sound signal into a light flashing formula to obtain a light flashing parameter value. According to the light flashing parameter value of the operation control unit, the left side lamps, the central lamps and the right side lamps respectively form staggered flashing, so that the flashing light set produces light changes corresponding to the music rhythm and vocals.

A method of using stereo recording to control the flashing of central lamps includes the steps of: providing left and right channel microphones in a stereo sound source venue or electronic device, and recording the plural sound sources generated by the left area, central area and right area of the stereo sound source, then storing the recorded analog signal in an audio signal sampling unit of a signal processing device, and then converting the sound signal into a light flashing formula to obtain a light flashing parameter value. According to the light flashing parameter value of the operation control unit, the left side lamps, the central lamps and the right side lamps respectively form staggered flashing, so that the flashing light set produces light changes corresponding to the music rhythm and vocals.

Voice data transmission with adaptive redundancy creates a voice data packet by packetizing the voice data payload and a number of redundant payloads selected from a set of previous voice data payloads. The voice data from the voice data payload is analysed to determine whether it is a critical or non-critical payload by classifying the received voice data as voiced or unvoiced. If at least a portion of the voice data is classified as unvoiced, the voice data payload is determined to be a critical payload. If it is a critical payload, then the voice data payload is added to the set of previous voice data payloads for inclusion as a redundant payload in subsequent voice data packets. The voice data packet is then forwarded for transmission over the network.

A system for producing an encoded digital audio recording has an audio encoder that encodes a digital audio recording having a number of audio channels or audio objects. An equalization (EQ) value generator produces a sequence of EQ values which define EQ filtering that is to be applied when decoding the encoded digital audio recording, wherein the EQ filtering is to be applied to a group of one or more of the audio channels or audio objects of the recording independent of any downmix. A bitstream multiplexer combines the encoded digital audio recording with the sequence of EQ values, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

A first playback device is configured to: operate as part of a synchrony group that comprises the first playback device and a second playback device; obtain a first version of audio content that is encoded according to a first encoding format; determine that the first version of the audio content is unsuitable for playback by the second playback device; based on the determination, (i) decode the first version of the audio content and (ii) re-encode a second version of the audio content according to a second encoding format; transmit the second version of the audio content to the second playback device for playback; cause the second playback device to play back the second version of the audio content; and play back the first version of the audio content in synchrony with the playback of the second version of the audio content by the second playback device.

Audiovisual presentations, such as film recordings, may have been originally created having an audio soundtrack with multiple audio tracks mixed for a surround sound system that includes a set of speakers physically surrounding a user. The present disclosure presents systems and methods to remix these soundtracks into 3D audio that when presented to the ears of a user can be perceived as a virtual surround sound system that mimics the physical system. What is more, the disclosed systems and methods can enhance the virtual surround sound system by adjusting virtual speakers of the virtual surround sound system according to video content of the audiovisual presentation. Further enhancement may be possible by adjusting the virtual speakers of the virtual surround sound system according to a sensed position of a user.

Disclosed speech recognition techniques improve user-perceived latency while maintaining accuracy by: receiving an audio stream, in parallel, by a primary (e.g., accurate) speech recognition engine (SRE) and a secondary (e.g., fast) SRE; generating, with the primary SRE, a primary result; generating, with the secondary SRE, a secondary result; appending the secondary result to a word list; and merging the primary result into the secondary result in the word list. Combining output from the primary and secondary SREs into a single decoder as described herein improves user-perceived latency while maintaining or improving accuracy, among other advantages.

The present disclosure generally relates to audio synchronization. An example method includes, at a first device with a communication device: performing an audio timing synchronization process that includes: transmitting, via the communication device, to a second device, a request for the second device to participate in the audio timing synchronization process; subsequent to transmitting the request to the second device, causing an output of an audio tone; subsequent to causing the output of the audio tone, receiving data from the second device based on the audio tone; and adjusting an audio timing synchronization setting of a third device based at least in part on the data received from the second device.