摘要:
An audio processing scheme is described. In an example, an apparatus comprises: at least two acoustic sensors through which audio content is received; at least one other sensor; an audio processor connected to the sensors and configured to receive audio information from the acoustic sensors and other information from the other sensor. The audio processor is configured to determine a use case of the apparatus based on the audio information and the other information. The audio processor is configured to adjust at least one audio processing scheme for the apparatus based on the determined use case. In other examples, a method and a computer program product are described.
摘要:
A device and method process voice communication service. A mobile terminal device of the present disclosure includes a microphone arranged at one end of a body of the device; a speaker arranged close to the microphone; a transceiver arranged at the other end of the body; a codec including a coder connected to the microphone, a decoder connected to the speaker, and a switch of which one node is connected to one of the coder and the decoder selectively and the other node is connected to the transceiver; and a communication controller which controls the switch to establish a path between the coder and the transceiver and enables the speaker in speakerphone mode.
摘要:
A device and method process voice communication service. A mobile terminal device of the present disclosure includes a microphone arranged at one end of a body of the device; a speaker arranged close to the microphone; a transceiver arranged at the other end of the body; a codec including a coder connected to the microphone, a decoder connected to the speaker, and a switch of which one node is connected to one of the coder and the decoder selectively and the other node is connected to the transceiver; and a communication controller which controls the switch to establish a path between the coder and the transceiver and enables the speaker in speakerphone mode.
摘要:
A system and method for dynamically establishing optimum audio quality in an audio conference is disclosed. A connection with one or more remote communication devices is initially established. An available data rate associated with the connection is then determined. Next, a bandwidth is assigned based on the available data rate. Finally, the assigned bandwidth is adjusted according to the available data rate.
摘要:
The present invention overcomes interface problems between proprietary handset ports on telephone base units and voice/data accessory products by allowing a user to automatically calibrate the telephone accessory product for an optimal interface match with the intended telephone base unit. This is accomplished through the use of a “Smart Interface Technology” (SIT) integrated chip set consisting of a full custom analog and semi-custom digital integrated circuit. The SIT incorporates three different methods for “learning” the characteristics of 4-wire port modular interfaces found in all telephone station sets. Basically, these methods determine the appropriate 4-wire terminal configurations, the transmit and receive channels of the intended telephone base unit, and adjust the channel sensitivities until an optimal and clear signal is provided for the user.
摘要:
A communication system has an audio receiver path, and a digital transceiver path, wherein both paths are integrated on a single integrated circuit. The audio receiver path has a concealment device, and the communication system also has at least one functional block connected to and shared by the both paths. The communication system has a digital controller connected to both paths, the at least one functional block, and the concealment device. The digital controller is operable to schedule the operation of the digital transceiver path, and to inform the concealment device of start and end points of the activity of the digital transceiver path. The concealment device is operable to mask interruptions in the audio stream from the audio receiver path caused during periods of activity of the digital transceiver path.
摘要:
A method for communicating data during an audio conference is provided. Digital data is received from a data source. The digital data is then modulated onto a carrier signal. The modulated carrier signal is subsequently combined with an audio signal of the audio conference. The combined signal is communicated to at least one remote communication device. Upon receipt by the at least one remote communication device, the combined signal is separated into the audio signal and the modulated carrier signal. The modulated carrier signal is then demodulated to extract the digital data. Both the digital data and the audio signal are output by the at least one remote communication device.
摘要:
A novel and useful apparatus for and method of integrating the advanced audio distribution profile (A2DP) audio codec into a Bluetooth controller for audio streaming applications. The mechanism functions to break the prior art Bluetooth protocol stack by integrating a profile packet composer into the controller. The profile/stack control signaling is performed by the host while the profile data packet composer is implemented in the controller. The integrated data packet composer does not break the data path and flow control over the standard HCI. Further, the integrated packet composer allows the controller to open a dedicated data interface for specific applications (e.g., PCM/I2S for audio data).
摘要:
A communication apparatus comprising an audio input device adapted to capture a first audio sample, where the first audio sample comprises a noise component. The apparatus further comprises signal processing logic coupled to the audio input device. If the intensity of the noise component is equal to or greater than the intensity of a voice component of a second audio sample received from a different communication apparatus, the signal processing logic amplifies the voice component.
摘要:
A mobile terminal for dynamically controlling delivery of active media during a call includes an output device and a processing element. The output device is capable of providing an audible output. The processing element is capable of delivering the active media to the output device of the mobile terminal while the call is on-going.