A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

Transmitting node and receiving node for audio coding and methods therein. The nodes being operable to encode/decode speech and to apply a discontinuous transmission (DTX) scheme comprising transmission/reception of Silence Insertion Descriptor (SID) frames during speech inactivity. The method in the transmitting node comprising determining, from amongst a number N of hangover frames, a set Y of frames being representative of background noise, and further transmitting the N hangover frames, comprising at least said set Y of frames, to the receiving node. The method further comprises transmitting a first SID frame to the receiving node in association with the transmission of the N hangover frames, where the SID frame comprises information indicating the determined set Y of hangover frames to the receiving node. The method enables the receiving node to generate comfort noise based on the hangover frames most adequate for the purpose.

Provided is an audio transcoding method, including: (301) performing entropy decoding on a first audio stream with a first bitrate, to obtain an audio feature parameter and an excitation signal of the first audio stream, the excitation signal being a quantized audio signal; (302) obtaining a time-domain audio signal corresponding to the excitation signal based on the audio feature parameter and the excitation signal; (303) re-quantizing the excitation signal and the audio feature parameter based on the time-domain audio signal and a target transcoding bitrate, to obtain a target excitation signal and a target audio feature parameter; and (304) performing entropy coding on the target audio feature parameter and the target excitation signal, to obtain a second audio stream with a second bitrate, the second bitrate being lower than the first bitrate.

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.

Transmitting node and receiving node for audio coding and methods therein. The nodes being operable to encode/decode speech and to apply a discontinuous transmission (DTX) scheme comprising transmission/reception of Silence Insertion Descriptor (SID) frames during speech inactivity. The method in the transmitting node comprising determining, from amongst a number N of hangover frames, a set Y of frames being representative of background noise, and further transmitting the N hangover frames, comprising at least said set Y of frames, to the receiving node. The method further comprises transmitting a first SID frame to the receiving node in association with the transmission of the N hangover frames, where the SID frame comprises information indicating the determined set Y of hangover frames to the receiving node. The method enables the receiving node to generate comfort noise based on the hangover frames most adequate for the purpose.

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

Methods, apparatus and articles of manufacture to identify sources of network streaming services are disclosed. An example apparatus includes a coding format identifier to identify, from a received first audio signal representing a decompressed second audio signal, an audio compression configuration used to compress a third audio signal to form the second audio signal, and a source identifier to identify a source of the second audio signal based on the identified audio compression configuration.

In some implementations, an application server may receive, from a calling party user equipment, a call for a called party associated with multiple user equipment. The application server may provide to the multiple user equipment, and based on the call, a request for transcoding information associated with the multiple user equipment. The application server may assign a transcoding resource for handling the call, wherein the transcoding resource is provided in a network. The application server may receive, based on the request, the transcoding information from a particular user equipment of the multiple user equipment. The application server may provide the transcoding information to the transcoding resource, wherein the transcoding information causes the transcoding resource to establish and transcode the call between the calling party user equipment and the particular user equipment.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

An electronic device and method thereof for outputting audio data, the electronic device including: an audio module for outputting at least one of a piece of first audio data having a designated format or a piece of second audio data having a format different from the designated format; a memory configured to store instructions; and a processor configured to execute the instructions to: mix (hereinafter, first-mix) the at least one piece of second audio data, convert the first-mixed audio data into the designated format, mix (hereinafter, second-mix) the at least one piece of first audio data and the audio data converted into the designated format, post-process the second-mixed audio data; and transmit the post-processed audio data to the audio module to be output through the first sound output device.