An apparatus for downmixing three or more audio input channels to obtain two or more audio output channels is provided. The apparatus includes a receiving interface for receiving the three or more audio input channels and for receiving side information. Moreover, the apparatus includes a downmixer for downmixing the three or more audio input channels depending on the side information to obtain the two or more audio output channels. The number of the audio output channels is smaller than the number of the audio input channels. The side information indicates a characteristic of at least one of the three or more audio input channels, or a characteristic of one or more sound waves recorded within the one or more audio input channels, or a characteristic of one or more sound sources which emitted one or more sound waves recorded within the one or more audio input channels.

In one embodiment, a method for transcript generation includes receiving an audio file and dividing it into a plurality of chunks. The method further includes sending each instance of the plurality of chunks to a speech service module. The method further includes converting speech to text for each instance of the plurality of chunks and returning the text for each instance of the plurality of chunks. The method further includes merging the text for each instance of the plurality of chunks to yield an audio file transcript and sending the audio file and chunks to a diarization module. The method further includes performing first pass diarization on the chunks to yield a plurality of diarized chunks and performing second pass diarization on the plurality of diarized chunks and the audio file to yield a diarized audio file. The method further includes merging the files to yield a final transcript.

Novel tools and techniques are provided for implementing split and parallelized encoding or transcoding of audio and video. In various embodiments, a computing system might split an audio-video file that is received from a content source into a single video file and a single audio file. The computing system might encode or transcode the single audio file. Concurrently, the computing system might split the single video file into a plurality of video segments. A plurality of parallel video encoders/transcoders might concurrently encode or transcode the plurality of video segments, each video encoder/transcoder encoding or transcoding one video segment of the plurality of video segments. Subsequently, the computing system might assemble the plurality of encoded or transcoded video segments with the encoded or transcoded audio file to produce an encoded or transcoded audio-video file, which may be output to a display device(s), an audio playback device(s), or the like.

The disclosure provides a speech processing method and a device thereof. The method includes: acquiring a speech sampling signal frame in a mixed-excitation linear prediction (MELP) speech coding system and estimating signal quality of the speech sampling signal frame; determining, based on the signal quality, a specific linear prediction coding (LPC) order used by an LPC circuit; controlling the LPC circuit to convert the speech sampling signal frame into a line spectrum pair parameter based on the specific LPC order; replacing a speech signal spectrum of the speech sampling signal frame with the line spectrum pair parameter to generate a predicted speech signal; and performing a speech coding operation and a signal synthesizing operation of the MELP speech coding system based on the predicted speech signal.

A device includes one or more processors configured to generate, at an encoder portion of an autoencoder, first output data at least partially based on first input data and to generate, at a decoder portion or the autoencoder, a representation of the first input data at least partially based on the first output data. The one or more processors are configured to generate, at the encoder portion, second output data based on second input data and first state data and to generate, at the decoder portion, a representation of the second input data based on the second output data and second state data. Each of the first state data and the second state data correspond to the state of the decoder portion resulting from generation of the representation of the first input data. The first and second input data correspond to sequential values of a signal to be encoded.

A communication node determines a codec and a codec mode to be used by two terminals that perform communication in a first network, when one of the two terminals performs handover to a second network that is different from the first network. The communication node includes: a determiner that sets common part for a codec and a codec mode to be used by the two terminals, the common part being common among information indicating codecs and codec modes used for communication in the first network, information indicating codecs and codec modes supported by the one terminal, and information indicating codecs and codec modes supported by the second network; and a generator that generates signaling for requesting the two terminals to perform changing to the set codec and codec mode to be used by the two terminals.

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.

An apparatus for processing an encoded audio signal, which includes a sequence of access units, each access unit including a core signal with a first spectral width and parameters describing a spectrum above the first spectral width, has a demultiplexer generating, from an access unit of the encoded audio signal, the core signal and a set of the parameters, an upsampler upsampling the core signal of the access unit and outputting a first upsampled spectrum and a timely consecutive second upsampled spectrum, the first upsampled spectrum and the second upsampled spectrum, both, having a same content as the core signal and having a second spectral width being greater than the first spectral width of the core spectrum, a parameter converter converting parameters of the set of parameters of the access unit to obtain converted parameters, and a spectral gap filling processor processing the first upsampled spectrum and the second upsampled spectrum using the converted parameters.

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.

A multi-channel signal decoding method includes obtaining a downmixed signal of a first channel signal and a second channel signal in a multi-channel signal, obtaining parameter indication information, obtaining a reverberation gain parameter corresponding to each of the part of subbands according to the parameter indication information; and obtaining a reconstructed first channel signal and a reconstructed second channel signal according to the downmixed signal and the reverberation gain parameter.