摘要:
This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.
摘要:
Embodiments of the present invention disclose a call method of a terminal and a terminal using the call method, relate to the field of communications technologies, and are invented for adjusting voice quality of a call in real time. The method includes: analyzing a spectral component of a voice signal during a call; and selecting a corresponding frequency response channel according to an analysis result of the spectral component of the voice signal. The present invention is mainly applied in the field of call services.
摘要:
Embodiments of the present invention provide a recording method, apparatus, and terminal, and relate to the audio and video recording and playing field. The present invention is invented to prevent a problem that a stereophonic effect of a recorded video is poor as a result of a change in location status of a terminal. The method includes: acquiring a placement status parameter of a terminal according to current placement status of the terminal; determining that one photographing unit of at least one photographing unit is in a started state; determining a recording correspondence between at least two audio sensor units and two sound channels according to the acquired placement status parameter and the determined photographing unit in the started state; and adjusting volume of the at least two audio sensor units according to the recording correspondence between the at least two audio sensor units and the two sound channels, so that a difference between volume that passes through the two sound channels is less than a preset threshold. The present invention may be applied to audio processing technologies.
摘要:
A method, apparatus and computer program product are provided to permit audio signals to provide additional information to a user regarding the distance to the source of the audio signals, thereby increasing a user's situational awareness. In the context of a method, a distance and a direction from a user to an object are determined. The method also scales the distance to the object to create a modified distance within a predefined sound field region about the user. The method also causes an audio cue relating to the object to be audibly provided to the user. The audio cue is such that the object appears to be located within the predefined sound field region in the direction and at the modified distance from the user from the user.
摘要:
A method, apparatus and computer program product are provided to permit audio signals to provide additional information to a user regarding the distance to the source of the audio signals, thereby increasing a user's situational awareness. In the context of a method, a distance and a direction from a user to an object are determined. The method also scales the distance to the object to create a modified distance within a predefined sound field region about the user. The method also causes an audio cue relating to the object to be audibly provided to the user. The audio cue is such that the object appears to be located within the predefined sound field region in the direction and at the modified distance from the user from the user.
摘要:
Embodiments of the present invention provide a method and a device for processing a sound signal for a communications device, where a relationship between values of a volume of a first sound signal collected by a main microphone and a volume of a second sound signal collected by an auxiliary microphone is acquired by comparison, to determine a sound signal processing policy; and according to the sound signal processing policy, a sound signal that is to be sent to a peer communications terminal is determined, where the sound signal processing policy is used to ensure that a volume of the sound signal that is to be sent to the peer communications terminal exceeds a preset volume threshold. According to the sound signal processing policy, a problem that in a communication process, a volume of voice heard by a user of a peer communications terminal is small or even no sound is heard is resolved.
摘要:
Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.