Patent application number | Description | Published |
20090012797 | Method and apparatus for encoding and decoding an audio signal using adaptively switched temporal resolution in the spectral domain - Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding/decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections. By applying a corresponding switching control, pre-echo effects are avoided and a high coding gain and a low coding delay are achieved. | 01-08-2009 |
20090106031 | Method and Apparatus for Re-Encoding Signals - At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction. | 04-23-2009 |
20090122191 | Method and Apparatus for Replaying a Video Signal and One or More Audio Signals Related to Audio/Video Data That are Based on a 24Hz Frame Frequency Video Signal - Movies are produced in 24 Hz frame frequency and progressive scanning format (denoted 24p) for projection in film theatres, adhering to a worldwide standard for 35 mm film. However, the major TV systems in the world use interlaced scanning and either 50 Hz field frequency (denoted 50i) or 60 Hz field frequency (denoted 60i). Content providers would prefer providing single-picture-frequency single-audio-speed AV discs that can be replayed in most parts of the world. According to the invention, For a 50 HZ output mode, in the media player either audio signal frames are dropped adaptively or video fields or frames are repeated adaptively, depending on the current video and audio content. Thereby the less perceptible stream controls the synchronisation. | 05-14-2009 |
20090122992 | Method and Apparatus for Encrypting Encoded Audio Signal - Advanced solutions for encrypting multi-layer audio data are required, ie. audio data that comprise a base layer and one or more enhancement layers. A method for encrypting such an encoded audio signal comprises separating the base layer into two sections, encrypting the side information within frames of the second section of the base layer, and encrypting at least a part of the data of the enhancement layer, wherein the encrypted section of the base layer and the encrypted enhancement layer require different decryption keys for decryption. Thus, free preview zones are possible to implement. | 05-14-2009 |
20090164226 | Method and Apparatus for Lossless Encoding of a Source Signal Using a Lossy Encoded Data Stream and a Lossless Extension Data Stream - In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding. | 06-25-2009 |
20090177478 | Method and Apparatus for Lossless Encoding of a Source Signal, Using a Lossy Encoded Data Steam and a Lossless Extension Data Stream - In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/decoding. | 07-09-2009 |
20090240506 | AUDIO BITSTREAM DATA STRUCTURE ARRANGEMENT OF A LOSSY ENCODED SIGNAL TOGETHER WITH LOSSLESS ENCODED EXTENSION DATA FOR SAID SIGNAL - Lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate, but not irrelevancies as identified by psycho-acoustics. Lossless audio coding schemes apply a filter or transform for decorrelation and then encode the transformed signal. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. However, in case of lossy based lossless coding the additional amount of information exceeds the amount of data for the base layer by a multiple of the base layer data amount. Therefore the additional data cannot be packed completely into the base layer data stream e.g. as ancillary data. The at least two data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. Furthermore several intermediate quality layers are possible. However, these data streams are not independent from each other Every higher layer depends on the lower layers and can only be reasonably decoded in combination with these lower layers. According to the invention, a special combination of one-time header information with repeated header information in a block structure is used, which kind of combination depends on the type of application. Assignment information data identify the different parts or layers of the lossless format belonging to one input signal. Synchronisation data are used to combine the different data streams or parts or layers to a single lossless or intermediate output signal. These features are used in a file format and in a streaming format. | 09-24-2009 |
20090306993 | METHOD AND APPARATUS FOR LOSSLESS ENCODING OF A SOURCE SIGNAL, USING A LOSSY ENCODED DATA STREAM AND A LOSSLESS EXTENSION DATA STREAM - The invention is related to lossless encoding of a source signal, using a lossy encoded data stream and a lossless extension data stream which together form a lossless encoded data stream for said source signal, whereby lossless audio compression means audio coding with bit-exact reproduction of the original PCM samples at decoder output. The lossy encoding/decoding may be an mp3 coding/decoding. The invention uses an integer MDCT and frequency domain de-correlation and time domain de-correlation for the residual signal of the base-layer lossy audio codec. The exploitation of side information from the lossy base-layer codec allows for reduction of redundancies in the gross bit stream, thus improving the coding efficiency of the lossy based lossless codec. | 12-10-2009 |
20110150097 | METHOD FOR ENCODING A BIT AMOUNT OF A DATE SECTION AN CORRESPONDING METHOD FOR DECODING, METHOD FOR ENCODING OR DECODING AUDIO AND/OR VIDEO DATA, METHOD FOR TRANSMITTING AUDIO AND/OR VIDEO DATA AND STORAGE MEDIUM COMPRISING AUDIO AND/OR VIDEO DATA - The invention relates to a method for encoding of a bit amount of a data section and to a corresponding decoding method. Furthermore, the invention relates to encoding, decoding, transmission and/or storage of audio and/or video data wherein said method for encoding of a bit amount of a data section and/or said corresponding decoding method are used in processing of the audio and/or video data. Said method for encoding of a bit amount of a data section comprises the steps of encoding said bit amount indicating integer as a first number of equally valued bits followed by a stop bit of different value wherein said first number equals said bit amount increased by a threshold value. Using said method, quotients of values larger than a threshold can be encoded using unary as well as binary code wherein quotients of values smaller than the threshold can be encoded in unary code. | 06-23-2011 |
20110158326 | METHOD AND APPARATUS FOR GENERATING OR CUTTING OR CHANGING A FRAME BASED BIT STREAM FORMAT FILE INCLUDING AT LEAST ONE HEADER SECTION, AND A CORRESPONDING DATA STRUCTURE - In frame-based bit stream formats the data required for decoding a current frame are usually stored within the data section for that frame. One exception is the mp3 bit stream where data for a current frame is stored in previous frames. If the decoder did not receive the required previous frame, decoding of the current mp3 frame is skipped. The invention can be applied for such bit streams, in an archival mode, a streaming mode and a sample-exact cutting of an archival mode. In the streaming and cutting modes, new headers are established. The number of frames required for initialising the decoder status is signalised in the header, as well as a consistency check value in the streaming mode. These frames are used for decoder initialisation but not for decoding samples or coefficients. For a sample-exact cutting, for the frame at which the cut shall occur, the number of samples or coefficients to be muted is also indicated in the header. The invention can be applied for the hd3 audio file format for lossless extension of an mp3 bit stream. | 06-30-2011 |
20110238424 | Method and apparatus for encoding and decoding excitation patterns from which the masking levels for an audio signal encoding and decoding are determined - For the quantisation of spectral data in an audio transform encoder psycho-acoustic information is required, i.e. an approximation of the true masking threshold. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window/transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied. | 09-29-2011 |
20120155653 | METHOD AND APPARATUS FOR ENCODING AND DECODING SUCCESSIVE FRAMES OF AN AMBISONICS REPRESENTATION OF A 2- OR 3-DIMENSIONAL SOUND FIELD - Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1) | 06-21-2012 |
20130010971 | METHOD AND DEVICE FOR DECODING AN AUDIO SOUNDFIELD REPRESENTATION FOR AUDIO PLAYBACK - Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics uses spherical harmonics of at least 2 | 01-10-2013 |
20130216070 | DATA STRUCTURE FOR HIGHER ORDER AMBISONICS AUDIO DATA - The invention is related to a data structure for Higher Order Ambisonics HOA audio data, which data structure includes 2D or 3D spatial audio content data for one or more different HOA audio data stream descriptions. The HOA audio data can have on order of greater than ‘3’, and the data structure in addition can include single audio signal source data and/or microphone array audio data from fixed or time-varying spatial positions. | 08-22-2013 |
20130236039 | METHOD AND APPARATUS FOR PLAYBACK OF A HIGHER-ORDER AMBISONICS AUDIO SIGNAL - An advantage of Ambisonics representation is that the reproduction of the sound field can be adapted individually to nearly any given loudspeaker position arrangement. The invention allows systematic adaptation of the playback of spatial sound field-oriented audio to its linked visible objects, by applying space warping processing as disclosed in EP 11305845.7. The reference size (or the viewing angle from a reference listening position) of the screen used in the content production is encoded and transmitted as metadata together with the content, or the decoder knows the actual size of the target screen with respect to a fixed reference screen size. The decoder warps the sound field in such a manner that all sound objects in the direction of the screen are compressed or stretched according to the ratio of the size of the target screen and the size of the reference screen. | 09-12-2013 |
20150081310 | METHOD AND APPARATUS FOR DECODING STEREO LOUDSPEAKER SIGNALS FROM A HIGHER-ORDER AMBISONICS AUDIO SIGNAL - Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localisation in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA. The desired panning functions can be derived from a panning law for placement of virtual sources between the loudspeakers. For each loudspeaker a desired panning function for all possible input directions at sampling points is defined. The panning functions are approximated by circular harmonic functions, and with increasing Ambisonics order the desired panning functions are matched with decreasing error. For the frontal region between the loudspeakers, a panning law like the tangent law or vector base amplitude panning (VBAP) are used. For the rear directions panning functions with a slight attenuation of sounds from these directions are defined. | 03-19-2015 |