Abstract:
A speech recognition system for resolving impaired utterances can have a speech recognition engine configured to receive a plurality of representations of an utterance and concurrently to determine a plurality of highest-likelihood transcription candidates corresponding to each respective representation of the utterance. The recognition system can also have a selector configured to determine a most-likely accurate transcription from among the transcription candidates. As but one example, the plurality of representations of the utterance can be acquired by a microphone array, and beamforming techniques can generate independent streams of the utterance across various look directions using output from the microphone array.
Abstract:
Method of improving audio signal in the spectral domain starts by receiving audio signal that includes signals from sources including speech source and music source. Audio signal is tuned for output by sound output device. Portions of audio signal are analyzed in a spectral domain to determine whether adjustments are required. Analyzing portions of audio signal includes determining whether anomaly is present in frequency band of audio signal in spectral domain by using at least one metric. Metrics include band energy ratios, spectral centroid, spectral tilt, spectral flux, spectral variance, absolute thresholds, and relative thresholds. Audio signal is adjusted to improve audio signal in spectral domain when audio signal is determined to require adjustments. Adjusting audio signal includes adjusting values of the metric in frequency band that is determined to include anomaly to correspond to clustering of metric values for audio signal in spectral domain. Other embodiments are also described.
Abstract:
A multi-band audio compressor that may provide not only better and brighter sound, but also speaker protection. The multi-band audio compressor breaks an input audio signal into different frequency bands. For each band signal, a volume re-mapper translates a user preference volume level to a converted volume level based on a programmable volume curve for the band signal. For each frequency band, the band signal is processed by a gain stage and a compressor. Each gain stage applies a signal gain to the band signal based on the converted volume level. Each compressor compresses the output of the gain stage. After compression, the different frequency band signals are re-combined and the combined audio signal may then be passed to a power amplifier that is driving a speaker. Other embodiments are also described and claimed.
Abstract:
A multi-band audio compressor that may provide not only better and brighter sound, but also speaker protection. The multi-band audio compressor breaks an input audio signal into different frequency bands. For each band signal, a volume re-mapper translates a user preference volume level to a converted volume level based on a programmable volume curve for the band signal. For each frequency band, the band signal is processed by a gain stage and a compressor. Each gain stage applies a signal gain to the band signal based on the converted volume level. Each compressor compresses the output of the gain stage. After compression, the different frequency band signals are re-combined and the combined audio signal may then be passed to a power amplifier that is driving a speaker. Other embodiments are also described and claimed.
Abstract:
Method of dynamically adapting user volume input range on mobile device having global volume range starts by receiving a volume input selection from a user that is level included in user volume input range. User volume input range is a portion of global volume range. Device's processor then detects ambient noise level surrounding device and adjusts user volume input range from current portion of global volume range to different portion of global volume range based on detected ambient noise level. Volume input selection remains at the same level included in user volume input range after user volume input range is adjusted. Processor may identify sound profile that corresponds to ambient noise level being detected and adjusts user volume input range to a different portion of the global volume range based on identified sound profile. Other embodiments are also described.
Abstract:
Improved systems and methods for psychoacoustic adaptive notch filtering are provided. By accounting for psychoacoustic properties of an audio signal as well as finer characteristics of noise which may be present in the audio signal (e.g., the shape of the spectral density of the noise), more effective strategies for dealing with undesirable components of the audio signal may be realized.