Abstract:
A computer-implemented method of encoding audio includes accessing a plurality of independent audio source streams, each of which includes a sequence of source frames. Respective source frames of each sequence include respective pluralities of pulse-code modulated audio samples. Each of the plurality of independent audio source streams is separately encoded to generate a plurality of independent encoded streams, each of which corresponds to a respective independent audio source stream. The encoding includes, for respective source frames, converting respective pluralities of pulse-code modulated audio samples to respective pluralities of floating-point frequency samples that are divided into a plurality of frequency bands. An instruction to mix the plurality of independent encoded streams is received; in response, respective floating-point frequency samples of the independent encoded streams are combined. An output bitstream is generated that includes the combined respective floating-point frequency samples.
Abstract:
Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. According to an example method, an identifier is encoded in media content when a different identifier has been previously encoded. According to another example method, messages decoded from media content are validated to provide improved decoding accuracy. In another example method, decoded symbols are stored in memory and synchronization symbols are located to detect a message encoded in media content.
Abstract:
This invention relates to reformatting a plurality of audio input signals from a first format to a second format by applying them to a dynamically-varying transformatting matrix. In particular, this invention obtains information attributable to the direction and intensity of one or more directional signal components, calculates the transformatting matrix based on the first and second rules, and applies the audio input signals to the transformatting matrix to produce output signals.
Abstract:
A vocoder and method transcodes Mixed Excitation Linear Prediction (MELP) encoded data for use at different speech frame rates. Input data is converte (100) into MELP parameters such as used by a first MELP vocoder. These parameters are buffered (102) and a time interpolation (104) is performed on the parameters with quantization to predict spaced points. An encoding function (106) is performed on the interpolated data as a block to produce a reduction in bit-rate as used by a second MELP vocoder at a different speech frame rate than the first MELP vocoder.
Abstract:
A vocoder and method transcodes Mixed Excitation Linear Prediction (MELP) encoded data for use at different speech frame rates. Input data is converte (100) into MELP parameters such as used by a first MELP vocoder. These parameters are buffered (102) and a time interpolation (104) is performed on the parameters with quantization to predict spaced points. An encoding function (106) is performed on the interpolated data as a block to produce a reduction in bit-rate as used by a second MELP vocoder at a different speech frame rate than the first MELP vocoder.
Abstract:
Conversion entre représentations en domaines de sous-bandes pour des bancs de filtres variant dans le temps L'invention concerne un traitement de transcodage entre domaines différents de sous- bandes, visant à compacter l'application d'un premier vecteur représentant le signal dans un premier domaine de sous-bandes à un banc de filtres de synthèse, puis à un banc de filtres d'analyse, pour obtenir un second vecteur représentant le signal dans un second domaine de sous-bandes. En particulier, le banc de synthèse et/ou le banc d'analyse sont variants dans le temps. On prévoit au sens de l'invention un filtrage matriciel du premier vecteur pour obtenir directement le second vecteur, ce filtrage matriciel étant représenté par une matrice globale de conversion comportant des sous-blocs matriciels (A i0 ,..., A ij ,..., A ip2-1 ) pré-calculés en tenant compte des variations possibles dans le temps des bancs de filtres, puis stockés en mémoire. La matrice globale de conversion est alors construite par appels à la mémoire pour obtenir les sous-blocs pré-calculés à des instants successifs.
Abstract:
A transcoder comprises a receiver (101) which receives input data representing an encoded signal and comprising first encoding data and first parametric extension data. The encoded data is fed to a decoder (103). The output of the decoder (103) is fed to an encoder (105) which generates second encoded data according to a different encoding protocol or with different encoding parameters. The first parametric extension data is fed to an extension data processor (109) which generates second parametric extension data directly from the first parametric extension data. The second encoded data and the second parametric extension data is combined in an output processor (107) to generate a transcoded signal comprising separately determined parametric extension data. The parametric extension data may be Spectral Band Replication (SBR) or Parametric Stereo (PS) extension data for an audio bitstream. Improved quality and reduced complexity is achieved by the separate transcoding of the parametric extension data.
Abstract:
A known time domain to frequency domain or frequency domain to time domain transform used in audio codecs is MDCT, which has the disadvantage of being costly in terms of required computational power due to high-precision multiplications, but which facilitates overlapping transform and subsampling. The invention uses a transform or inverse transform which does not involve multiplications because the transform and inverse transform matrices include '+1' and '-1' values only, but whereby the advantages of overlapping and subsampling are kept.