An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first and second encoding branches, the second encoding branch having a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and the second encoding branch having a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder, and a third domain decoder as well as two cascaded switches for switching between the decoders.

Present disclosure disclose a voice signal processing method, includes: receiving a first voice coded signal from a first terminal; performing voice decoding processing on the first voice coded signal to obtain a voice decoding parameter and a first voice decoded signal; performing, by using the voice decoding parameter, virtual bandwidth extension processing to obtain a bandwidth extension voice decoded signal corresponding to the first voice decoded signal; after combining the first voice decoded signal and the bandwidth extension voice decoded signal, performing voice coding processing to obtain a second voice coded signal; and sending the second voice coded signal to a second terminal that establishes a call connection to the first terminal, where a maximum frequency bandwidth supported by the first terminal is less than a maximum frequency bandwidth supported by the second terminal. Thus, Service quality of terminals that have asymmetric maximum frequency bandwidth support capabilities can be improved.

Example methods and apparatus to audio watermarking and watermark detection and extraction are disclosed herein. An example apparatus disclosed herein includes memory, computer readable instructions, and processor circuitry to execute the computer readable instructions to at least detect a first symbol, a second symbol, a third symbol, and a fourth symbol sequentially in encoded audio samples, determine whether the first symbol is a synchronization symbol, in response to a determination that the first symbol is a synchronization symbol, determine that the first symbol and the third symbol are associated with a first message and the second symbol and the fourth symbol are associated with a second message, and output at least one of the first message or the second message.

An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by a writer of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are set for each of playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data with the second format.

The disclosure discloses a system and method for real-time adjustment of a volume during live broadcasting, the system being arranged in a live broadcasting backend, wherein the system includes a transcoder and a volume adjusting device, wherein the transcoder includes a decoding unit and an encoding unit, wherein the decoding unit is configured to decode in real time a live broadcasting audio and video uploaded to the live broadcasting backend into original audio and video signals; and the encoding unit is configured to encode in real time the original audio and video signals into encoded audio and video signals; and the volume adjusting device is arranged between the decoding unit and the encoding unit, and configured to adjust the volume of the original audio signal output by the decoding unit to the encoding unit, in response to a volume adjusting instruction.

The present disclosure discloses method, apparatus, and a terminal for playing multimedia content, and relates to the field of computer technologies. The method includes: identifying a video encoding format, an audio encoding format, and a file format of a multimedia file; detecting whether the video encoding format can be processed by a graphics processing unit; when the video encoding format cannot be processed by the GPU, playing the multimedia file by using an application program that can process the video encoding format through a central processing unit; when at least one of the audio encoding format and the file format cannot be processed by the GPU, transcoding the multimedia file from the at least one format into a corresponding format that can be processed by the GPU; and sending the transcoded multimedia file to the GPU, so that the GPU plays the multimedia file.

This disclosure describes a virtual multipoint control unit in communication with one or more unified communication (UC) applications on one or more host devices over a network that allows ad hoc UC conferences between UC applications from different vendors. The virtual multipoint control unit includes a host device coupled to a network. In addition, the unit includes one or more unified communication (UC) applications being executed in software instructions. The virtual multipoint control unit virtualizes the physical audio devices to be shared simultaneously between multiple UC applications and allows ad hoc UC conferences between said UC applications from the same and or different UC vendors.

The present technology relates to an encoding device and method, a decoding device and method, and a program which can obtain high quality audio with a less code amount. A signal encoding unit encodes audio signals and outputs a resultant signal code string. A coefficient encoding unit encodes mixing coefficients for use in a mixing process of the audio signals and outputs a resultant coefficient code string. A multiplexing unit multiplexes the signal code string and the coefficient code string and outputs a resultant output code string. The coefficient encoding unit rearranges the mixing coefficients at the time of encoding the mixing coefficients on the basis of distances between input-side sound source positions and speaker positions on a reproduction side and calculates differential values between the mixing coefficients on the basis of arrangement order of the mixing coefficients, thereby encoding the mixing coefficients. The present technology is applicable to the encoding device and the decoding device.

An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system (500) comprises a buffer (510) and a decoding section (520). The buffer joins sets of audio data (D.sub.1; D.sub.2, . . . , D.sub.N) carried by N respective frames (F.sub.1, F.sub.2, . . . , F.sub.N) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Disclosed herein is an audio transmitter receiver device. The device includes an audio interface providing an audio signal, the audio signal including at least one of an audio input signal and an audio output signal; a digital communications interface for at least communicating audio information; and an audio codec for transcoding the audio information such that the audio information includes at least a high quality distortion free lossless representation of the audio signal and the audio signal includes an audio representation of the audio information.