Lossless Audio Signal Coding; Perfect Reconstruction Of Coded Audio Signal By Transmission Of Coding Error (epo) Patents (Class 704/E19.004)
  • Publication number: 20110106546
    Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.
    Type: Application
    Filed: March 9, 2010
    Publication date: May 5, 2011
    Inventor: Zoran Fejzo
  • Publication number: 20110060594
    Abstract: An audio encoder capable of implementing a plurality of encoding functions, wherein an adaptation controller adjusts the implementation of the encoding functions in response to feedback received by the adaptation controller during use. The adjustment may involve adapting encoding algorithms or selecting alternative encoding algorithms. The encoder may also include an operations scheduler to adjust the order in which the encoding functions are applied. The feedback may be received from internally of the encoder, for example from the currently implemented encoding functions, or from externally of the encoder. A corresponding decoder is also provided.
    Type: Application
    Filed: November 16, 2009
    Publication date: March 10, 2011
    Applicant: APT LICENSING LIMITED
    Inventor: David Trainor
  • Patent number: 7876905
    Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.
    Type: Grant
    Filed: May 8, 2007
    Date of Patent: January 25, 2011
    Inventor: Terry D. Beard
  • Publication number: 20100332238
    Abstract: A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L?1 remaining possible distinct data values, wherein each of the L?1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L?1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L?1 remaining possible distinct data values in the N samples, and encoding the L?1 coding vectors.
    Type: Application
    Filed: June 18, 2010
    Publication date: December 30, 2010
    Inventors: Lorin Paul Netsch, Jacek Piotr Stachurski
  • Publication number: 20100223052
    Abstract: A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a target band filter to form a regenerated speech signal in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal, the method comprising the step of controlling the modulated samples to lie in a second range of frequencies identified by determining a signal characteristic of frequencies in the first range of frequencies.
    Type: Application
    Filed: December 10, 2009
    Publication date: September 2, 2010
    Inventors: Mattias Nilsson, Soren Vang Anderson, Koen Bernard Vos
  • Publication number: 20100198603
    Abstract: A sub-band processing system that reduces computational complexity and memory requirements includes a processor and a local or distributed memory. Logic stored in the memory partitions a frequency spectrum of bins into a smaller number of sub-bands. The logic enables a lossy compression by designating a magnitude and a designated or derived phase of each bin in the frequency spectrum as representative. The logic renders a lossless compression by decompressing the lossy compressed data and providing lost data based on original spectral relationships contained within the frequency spectrum.
    Type: Application
    Filed: January 29, 2010
    Publication date: August 5, 2010
    Applicant: QNX SOFTWARE SYSTEMS(WAVEMAKERS), Inc.
    Inventor: Shreyas Paranjpe
  • Publication number: 20100082352
    Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.
    Type: Application
    Filed: November 5, 2009
    Publication date: April 1, 2010
    Inventor: Zoran Fejzo
  • Publication number: 20100063827
    Abstract: A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal.
    Type: Application
    Filed: September 4, 2009
    Publication date: March 11, 2010
    Applicant: GH Innovation, Inc.
    Inventor: Yang Gao
  • Publication number: 20100017196
    Abstract: Embodiments of methods, apparatuses, devices and systems associated with compression and decompression of digital signals are disclosed.
    Type: Application
    Filed: July 16, 2009
    Publication date: January 21, 2010
    Applicant: QUALCOMM Incorporated
    Inventors: Sang-Uk Ryu, Samir K. Gupta, Philip Rutschman, Heejong Yoo
  • Publication number: 20090306993
    Abstract: 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.
    Type: Application
    Filed: July 12, 2007
    Publication date: December 10, 2009
    Applicant: THOMSON LICENSING
    Inventors: Oliver Wuebbolt, Florian Keiler, Peter Jax, Sven Kordan, Johannes Boehm
  • Publication number: 20090177478
    Abstract: 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.
    Type: Application
    Filed: April 18, 2007
    Publication date: July 9, 2009
    Applicant: THOMSON LICENSING
    Inventors: Peter Jax, Florian Keiler, Oliver Wuebbolt, Sven Kordon, Johannes Boehm
  • Publication number: 20090164226
    Abstract: 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.
    Type: Application
    Filed: April 18, 2007
    Publication date: June 25, 2009
    Inventors: Johannes Boehm, Peter Jax, Florian Keiler, Oliver Wuebbolt, Sven Kordon
  • Publication number: 20090164223
    Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.
    Type: Application
    Filed: December 19, 2007
    Publication date: June 25, 2009
    Inventor: Zoran Fejzo
  • Publication number: 20090164224
    Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.
    Type: Application
    Filed: December 19, 2007
    Publication date: June 25, 2009
    Inventor: Zoran Fejzo
  • Publication number: 20090063140
    Abstract: An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.
    Type: Application
    Filed: October 31, 2005
    Publication date: March 5, 2009
    Applicant: KONINKLIJKE PHILIPS ELECTRONICS, N.V.
    Inventors: Lars Falck Villemoes, Erik Gosuinus Petrus Schuijers
  • Publication number: 20090012797
    Abstract: 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.
    Type: Application
    Filed: June 4, 2008
    Publication date: January 8, 2009
    Inventors: Johannes Boehm, Sven Kordon
  • Publication number: 20090012796
    Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a compatible down-mix signal optimized for a first multi-channel decoder from the input bitstream, converting the compatible down-mix signal to be optimized for a second multi-channel signal by performing a compatibility processing operation on the compatible down-mix signal, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the converted down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.
    Type: Application
    Filed: February 7, 2007
    Publication date: January 8, 2009
    Applicant: LG ELECTRONICS INC.
    Inventors: Yang Won Jung, Hee Suk Pang, Hyen O Oh, Dong Soo Kim, Jae Hyun Lim
  • Publication number: 20080300868
    Abstract: A digital recording apparatus for recording 1-bit digital audio data of a first sampling frequency on a recording medium in accordance with the recording format of multi-bit PCM data of a second sampling frequency includes a storage section to which input 1-bit digital audio data of the first sampling frequency is written; an encoder configured to read, from the storage section, the 1-bit digital audio data at a clock synchronized with the second sampling frequency and configured to convert the 1-bit digital audio data in such a manner that bits of the 1-bit digital audio data are arrayed in a 1-bit data area provided in the multi-bit PCM data that is in accord with the recording format; and a recorder configured to record data output from the encoder on the recording medium in accordance with the recording format.
    Type: Application
    Filed: May 29, 2008
    Publication date: December 4, 2008
    Inventor: Shinya Okada
  • Publication number: 20080215317
    Abstract: A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point.
    Type: Application
    Filed: January 30, 2008
    Publication date: September 4, 2008
    Inventor: Zoran Fejzo
  • Publication number: 20080133247
    Abstract: Disclosed is a method in a network element of a communication network, which communication network is capable of transparently transferring coded data at least in some part of the communication network. The method includes detecting a need to change codec rate to a second codec rate in a downlink connection from the communication network to an end user device; receiving coded data destined to said end user device, which data is coded with a first codec rate, and starting, in response to said detecting, rate transformation for transforming codec rate of said data destined to the end user device into said second codec rate. Also disclosed are an apparatus, a system and a computer program.
    Type: Application
    Filed: December 5, 2006
    Publication date: June 5, 2008
    Inventor: Antti Kurittu
  • Publication number: 20080021712
    Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.
    Type: Application
    Filed: August 14, 2007
    Publication date: January 24, 2008
    Inventor: Zoran Fejzo