Dolby International AB Patent Group Europe

Described are methods of processing an audio signal for packet loss concealment. The audio signal comprises a sequence of frames, each frame containing representations of a plurality of audio channels and reconstruction parameters for upmixing the plurality of audio channels to a predetermined channel format. One method includes: receiving the audio signal; and generating a reconstructed audio signal in the predefined channel format based on the received audio signal. Generating the reconstructed audio signal comprises: determining whether at least one frame of the audio signal has been lost; and if a number of consecutively lost frames exceeds a first threshold, fading the reconstructed audio signal to a predefined spatial configuration. Also described is a method of encoding an audio signal. Yet further described are apparatus for carrying out the methods, as well as corresponding programs and computer-readable storage media.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

The present document relates to time-alignment of encoded data of an audio encoder with associated metadata, such as spectral band replication (SBR) metadata. An audio decoder (100, 300) configured to determine a reconstructed frame of an audio signal (237) from an access unit (110) of a received data stream is described. The access unit (110) comprises waveform data (111) and metadata (112), wherein the waveform data (111) and the metadata (112) are associated with the same reconstructed frame of the audio signal (127). The audio decoder (100, 300) comprises a waveform processing path (101, 102, 103, 104, 105) configured to generate a plurality of waveform subband signals (123) from the waveform data (111), and a metadata processing path (108, 109) configured to generate decoded metadata (128) from the metadata (111).

A user prompt for a diffusion model to generate an image is received. Noisy latent features generated by denoising layers of the diffusion model are collected based on the user prompt and are to generate a pre-customized version of the image. Pre-customized lighting properties of the pre-customized version of the image are predicted based on the noisy latent features. Light source properties of a light source to be logically applied to generate a customized version of the image are determined. The light source properties of the light source are applied to the pre-customized lighting properties of the pre-customized version of the image to generate customized lighting properties of the customized version of the image. A loss computed from the customized lighting properties and the pre-customized lighting properties is used to guide the diffusion model to generate the customized version of the image.

The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Methods, apparatus, programs, and storage media for enhancing speech signals based on a neural network are provided. The method includes receiving a speech signal and a corresponding reference speech signal. A first set of band features is extracted from the speech signal and a second set of band features is extracted from the reference speech signal, wherein each feature in the first set of band features corresponds to a frequency band of the speech signal and each feature in the second set of band features corresponds to a frequency band of the reference speech signal. The first set of band features is split into a low frequency band part and high frequency band part. Band features corresponding to the high frequency band part are mapped to a reduced number of high frequency band features. The low frequency band part and the reduced number of high frequency band features are combined into a third set of band features. A first mask is determined by inputting the third set of band features into a speech enhancement neural subnetwork. The first mask is split into a low frequency mask part and high frequency mask part. Entries corresponding to the high frequency mask part are mapped to an increased number of entries. The low frequency mask part and the increased number of entries are combined into a second mask. A fourth set of band features is determined based on the second mask and the speech signal. A loss function is evaluated based on the second set of band features and the fourth set of band features. Weights of the speech enhancement neural subnetwork are updated based on a value of the evaluated loss function.

An audio processing unit, comprising a buffer memory (201) configured to store at least one frame of an encoded audio bitstream which includes audio data and a metadata container having a header and one or more metadata payloads after the header, a parser (205) coupled to the buffer memory, and an audio decoder (202) coupled to the parser. The one or more metadata payloads include dynamic range compression (DRC) metadata including DRC control values indicative of a compression profile assumed by an encoder to generate said DRC control values. The audio decoder is configured to perform dynamic range compression on decoded audio data generated by decoding said at least some of the audio data, using the DRC metadata.

The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.