Abstract: Various exemplary aspects are directed to acoustic context recognition apparatuses and methods involving isolating and identifying context(s) of an acoustic environment. In one exemplary embodiment, source audio is converted into audio spectrograms, each spectrogram indicative of a period of time. The series of spectrograms are analyzed to identify audio patterns, over a period of time, which are indicative of an environmental context of the source audio. In many embodiments of the present disclosure, acoustic context recognition also includes comparing the identified audio patterns to known environmental contexts.

Abstract: Embodiments of the invention include a device for reducing noise. The device may include a storage configured to store noise data; a processor configured to: classify a segment of noise utilizing noise data which was accumulated prior to initiation of a communication session; estimate the segment of noise, utilizing information received from the noise classification; and select a noise profile which accounts for a user's current context based on a context defined by the data which was accumulated prior to initiation of the communication session.

Abstract: Various exemplary aspects are directed to acoustic context recognition apparatuses and methods involving isolating and identifying context(s) of an acoustic environment. In one exemplary embodiment, source audio is converted into audio spectrograms, each spectrogram indicative of a period of time. The series of spectrograms are analyzed to identify audio patterns, over a period of time, which are indicative of an environmental context of the source audio. In many embodiments of the present disclosure, acoustic context recognition also includes comparing the identified audio patterns to known environmental contexts.

Abstract: Embodiments of the invention include a device for reducing noise. The device may include a storage configured to store noise data; a processor configured to: classify a segment of noise utilizing noise data which was accumulated prior to initiation of a communication session; estimate the segment of noise, utilizing information received from the noise classification; and select a noise profile which accounts for a user's current context based on a context defined by the data which was accumulated prior to initiation of the communication session.

Abstract: Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec.

Abstract: A system and method for discarding or inserting audio frames in a jitter buffer is described. The system and method provides improved audio quality as compared to conventional jitter buffer management systems. In one embodiment, buffer control logic determines whether to discard audio frames to be stored in a jitter buffer or to insert audio frames among audio frames to be output from a jitter buffer based not only on the number of audio frames currently stored in the jitter buffer but also based on the power of the current audio frame to be stored in or output from the jitter buffer. The system and method is generally applicable to any wireless or wired communication system in which audio signals are transmitted between entities operating in different clock domains.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A system and method for providing an augmented version of a Low-Complexity Sub-band Coder (LC-SBC) is described herein. In accordance with the method, a series of input audio samples representative of the frame are received. A series of sub-band samples is generated for each of a plurality of frequency sub-bands based on the input audio samples. A determination is made as to whether the frame is a voice frame or a noise frame. Responsive to a determination that the frame is a noise frame, an index representative of a previously-processed series of sub-band samples stored in a history buffer for at least one of the frequency sub-bands is encoded instead of encoding the series of sub-band samples generated for the frequency sub-band.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a wireless link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A system and method for discarding or inserting audio frames in a jitter buffer is described. The system and method provides improved audio quality as compared to conventional jitter buffer management systems. In one embodiment, buffer control logic determines whether to discard audio frames to be stored in a jitter buffer or to insert audio frames among audio frames to be output from a jitter buffer based not only on the number of audio frames currently stored in the jitter buffer but also based on the power of the current audio frame to be stored in or output from the jitter buffer. The system and method is generally applicable to any wireless or wired communication system in which audio signals are transmitted between entities operating in different clock domains.

Abstract: A system and method is described for updating the state of an audio decoder, such as a CVSD decoder, after a packet loss has occurred. In response to the loss of a packet, the system and method encodes audio samples produced by a packet loss concealment (PLC) algorithm and effectively passes the encoded audio samples through the audio decoder in lieu of the contents of the lost packet. This operation brings the state of the audio decoder into better synchronization with the state of a remote audio encoder, thereby reducing or minimizing the degrading effect of the packet loss on the perceived quality of an output audio signal produced by a voice processing system that includes the audio decoder.

Abstract: A system and method for discarding or inserting audio frames in a jitter buffer is described. The system and method provides improved audio quality as compared to conventional jitter buffer management systems. In one embodiment, buffer control logic determines whether to discard audio frames to be stored in a jitter buffer or to insert audio frames among audio frames to be output from a jitter buffer based not only on the number of audio frames currently stored in the jitter buffer but also based on the power of the current audio frame to be stored in or output from the jitter buffer. The system and method is generally applicable to any wireless or wired communication system in which audio signals are transmitted between entities operating in different clock domains.

Abstract: Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a wireless link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: An augmented version of a Low-Complexity Sub-band Coder (LC-SBC) is described herein that is better suited than conventional LC-SBC for wideband voice communication in the Bluetooth™ framework, where minimizing the power consumption is of paramount importance. The augmented version of LC-SBC utilizes certain techniques associated with speech coding, such as Voice Activity Detection (VAD), to reduce bandwidth usage and power consumption while maintaining voice quality. The augmented version of LC-SBC reduces the average bit rate used for transmitting wideband speech in a manner that does not add significant computational complexity. Furthermore, the augmented version of LC-SBC may advantageously be implemented in a manner that does not require any modification of the underlying logic/structure of LC-SBC.

Abstract: A system and method for discarding or inserting audio frames in a jitter buffer is described. The system and method provides improved audio quality as compared to conventional jitter buffer management systems. In one embodiment, buffer control logic determines whether to discard audio frames to be stored in a jitter buffer or to insert audio frames among audio frames to be output from a jitter buffer based not only on the number of audio frames currently stored in the jitter buffer but also based on the power of the current audio frame to be stored in or output from the jitter buffer. The system and method is generally applicable to any wireless or wired communication system in which audio signals are transmitted between entities operating in different clock domains.

Abstract: In a Bluetooth™ Sub-band Codec (SBC), the size of a bit pool allocated for encoding is adapted in a manner that is dependent upon the audio content that is being encoded. In one implementation, the size of the bit pool is increased during periods when the audio input signal represents an active audio signal, such as speech or music, and decreased when the audio input signal represents background noise or silence. This has the effect of increasing the bit rate (and thus the audio quality) in the presence of speech or music but decreasing the bit rate (and thus the audio quality) in the presence of background noise or silence. By adapting the size of the bit pool in this manner, the quality of the encoded bit-stream transmitted from the SBC encoder may be improved or power may be conserved depending upon the implementation.