A broadcast system may have less restrictive timing constraints by providing synchronous processing chains for the HD audio portion and the FM audio portion so that no samples are added or removed from when the input audio is first sampled at an input rate and when the signals are combined and output by a digital analog converter operating at an output rate. The signals can be buffered within the synchronous processing paths and the state of the buffer can be used to control the input rate of the sampler. Graceful change over across multiple input sources can be achieved provided all input source input rates are controlled to by the overall output rate and all input sources are phase aligned to produce output symbols at the same time.

An audio signal encoding method includes obtaining a current frame of an audio signal, where the current frame includes a high frequency band signal and a low frequency band signal; obtaining a parameter of bandwidth extension of the current frame based on the high frequency band signal, the low frequency band signal, and configuration information of the bandwidth extension; obtaining tile information, where the tile information indicates a first frequency range in which tonal component detection needs to be performed on the high frequency band signal; performing tonal component detection in the first frequency range to obtain information about a tonal component of the high frequency band signal; and performing bitstream multiplexing on the parameter of the bandwidth extension and the information of the tonal component to obtain a payload bitstream.

An audio encoding bit rate prediction model training method is performed by a computer device. The method includes: obtaining a sample audio feature parameter corresponding to each of sample audio frames in a first sample audio; performing encoding bit rate prediction on the sample audio feature parameter through an encoding bit rate prediction model, to obtain a sample encoding bit rate for each of the sample audio frames; performing audio encoding on the sample audio frames based on the corresponding sample encoding bit rates to generate sample audio data corresponding to the sample audio frames; performing audio decoding on the sample audio data, to obtain a second sample audio corresponding to the sample audio data; and training the encoding bit rate prediction model based on the first sample audio and the second sample audio until a sample encoding quality score reaches a target encoding quality score.

An audio signal encoding method performed by an encoder includes identifying an audio signal of a time domain in units of a block, generating a combined block by combining i) a current original block of the audio signal and ii) a previous original block chronologically adjacent to the current original block, extracting a first residual signal of a frequency domain from the combined block using linear predictive coding of a time domain, overlapping chronologically adjacent first residual signals among first residual signals converted into a time domain, and quantizing a second residual signal of a time domain extracted from the overlapped first residual signal by converting the second residual signal of the time domain into a frequency domain using linear predictive coding of a frequency domain.

Disclosed is an apparatus and method for encoding/decoding an audio signal using information of a previous frame. An audio signal encoding method includes: generating a current latent vector by reducing dimension of a current frame of an audio signal; generating a concatenation vector by concatenating a previous latent vector generated by reducing dimension of a previous frame of the audio signal with the current latent vector; and encoding and quantizing the concatenation vector.

According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

An encoder for encoding secondary media data including metadata and control data for primary media data is shown, wherein the encoder is configured to encode the secondary media data using adding redundancy or bandlimiting and wherein the encoder is configured to output the encoded secondary media data as a stream of digital words. Therefore, the stream of digital words may be formed such that it is capable to resist a typical processing of a digital audio stream. Furthermore, processors for processing a digital audio stream are able to process the stream of digital words, since the stream of digital words may be designed as an audio-like or analog-like digital stream.

It is attempted to reduce the processing load of a receiver at the time of integrating plural audio streams.

A predetermined number of audio streams are generated, and a container of a predetermined format including these predetermined number of audio streams is transmitted. The audio streams are constituted by an audio frame including a first packet that includes encoded data as payload information and a second packet that includes configuration information representing a configuration of the payload information of this first packet as payload information. Common index information is inserted in payloads of related first packet and second packet.

Systems and methods are presented for cross-fading (or other multiple clip processing) of information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Multiple clip processing can be accomplished at a client end according to directions sent from a service provider that specify a combination of (i) the clips involved; (ii) the device on which the cross-fade or other processing is to occur and its parameters; and (iii) the service provider system. For example, a consumer device with only one decoder, can utilize that decoder (typically hardware) to decompress one or more elements that are involved in a cross-fade at faster than real time, thus pre-fetching the next element(s) to be played in the cross-fade at the end of the currently being played element. The next elements(s) can, for example, be stored in an input buffer, then decoded and stored in a decoded sample buffer, all prior to the required presentation time of the multiple element effect. At the requisite time, a client device component can access the respective samples of the decoded audio clips as it performs the cross-fade, mix or other effect. Such exemplary embodiments use a single decoder and thus do not require synchronized simultaneous decodes.

An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.