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: 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: A device provides user interfaces for capturing and sending media, such as audio, video, or images, from within a message application. The device detects a movement of the device and in response, plays or records an audio message. The device sends the recorded audio message in response to detecting a movement of the device. The device removes messages from a conversation based on expiration criteria. The device shares a location with one or more message participants in a conversation.

Abstract: An equalizer that linearly interpolates between two equalization states when transitioning from one equalization state to the other equalization state is described. The equalizer includes a transfer function generator and an equalization module. Each equalization state is defined or determined based on a set of parameters. The transfer function generator generates a set of interpolated transfer functions by performing linear interpolation on a first equalization state and a second equalization state based on the set of parameters. The linear interpolation is performed on corresponding Z-domain poles and zeros of the transfer functions of the first and second equalization states. The equalization module applies the set of interpolated transfer functions generated by the transfer function generator to an input audio signal.

Abstract: Automatic gain control systems disclosed herein can incorporate a confidence metric that can estimate the accuracy of gain adjustments calculated by an automatic gain control module. The confidence metric may be based on a percentage of valid audio samples in a given period of time. Based on the confidence metric, the AGC response may be reduced, delayed, frozen, or otherwise altered from the baseline gain adjustment. Time-averaging process may be used to estimate the input signal power level and determine an appropriate baseline gain adjustment. Additionally, weighting functions can be adjusted to prevent overestimation of the signal power.

Abstract: Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

Abstract: Apparatuses, methods, computer readable mediums, and systems are described for combined dynamic processing and speaker protection for minimizing distortion in audio playback. In some embodiments, at least one compressed audio signal is received, at least one threshold for a speaker is retrieved, modifications to audio signal compression are determined based on the at least one compressed audio signal and the at least one threshold, information embodying the modifications is transmitted to a dynamic processor, and using the dynamic processor, at least one modified compressed audio signal is produced for the speaker based on the information.

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.

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.