Abstract: Methods, apparatus, and systems are disclosed to segment audio and determine audio segment similarities. An example apparatus includes at least one memory storing instructions and processor circuitry to execute instructions to at least select an anchor index beat of digital audio, identify a first segment of the digital audio based on the anchor index beat to analyze, the first segment having at least two beats and a respective center beat, concatenate time-frequency data of the at least two beats and the respective center beat to form a matrix of the first segment, generate a first deep feature based on the first segment, the first deep feature indicative of a descriptor of the digital audio, and train internal coefficients to classify the first deep feature as similar to a second deep feature based on the descriptor of the first deep feature and a descriptor of a second deep feature.

Abstract: A single-bit audio stream can be converted to a modified single-bit audio stream with a constant edge rate while maintaining a modulation index of the original audio stream using direct mapping. With direct mapping, a pre-filter bank may be combined with a multi-bit symbol mapper to select symbols for the modified audio stream with a constant edge rate per symbol and the same modulation index as the original audio stream. The output of the pre-filter bank may be an audio stream with no consecutive full-scale symbols. Using the output of the pre-filter bank, a multi-bit symbol mapper may use the symbol selector to output a symbol with a constant edge rate per symbol and the same modulation index as the original signal. The symbols may be converted to an analog signal for reproduction of audio content using a transducer.

Abstract: A stereo signal encoding method includes obtaining a residual signal encoding parameter of a current frame of a stereo signal based on downmixed signal energy and residual signal energy of each of M sub-bands of the current frame, where the residual signal encoding parameter indicates whether to encode residual signals of the M sub-bands, determining whether to encode the residual signals based on the residual signal encoding parameter, and encoding the residual signals when it is determined that the residual signals need to be encoded.

Abstract: A method includes receiving a non-linguistic input associated with an input musical content. The method also includes, using a model that embeds multiple musical features describing different musical content and relationships between the different musical content in a latent space, identifying one or more embeddings based on the input musical content. The method further includes at least one of: (i) identifying stored musical content based on the one or more identified embeddings or (ii) generating derived musical content based on the one or more identified embeddings. In addition, the method includes presenting at least one of: the stored musical content or the derived musical content. The model is generated by training a machine learning system having one or more first neural network components and one or more second neural network components such that embeddings of the musical features in the latent space have a predefined distribution.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Abstract: An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.

Abstract: A circuit is configured to reduce a noise component of a measured phase signal. The circuit includes an input for a phase signal of an oscillator and an error signal estimator configured to determine parity information and an estimated error amplitude in the phase signal based on the parity information. The circuit further includes a combiner configured to provide the measured phase signal with the reduced noise component based on a combination of the phase signal and the estimated error amplitude.

Abstract: An audio and video synchronizing perceptual model is described that is based on how a person perceives audio and/or video (e.g., how the brain processes sound and/or visual content). The relative emotional impact associated with different audio portions may be employed to determine transition points to facilitate automatic synchronization of audio data to video data to create a production that achieves a particular overall emotional effect on the listener/viewer. Various processing techniques of the perceptual model may utilize perceptual characteristics within the audio portions to determine a transition point for automatic synchronization with video data.

Abstract: Aspects disclosed in the detailed description include packetizing encoded audio frames into compressed-over-pulse code modulation (PCM) (COP) packets for transmission over PCM interfaces. In one aspect, a COP packetizing circuit is configured to receive an encoded audio frame generated from a PCM frame, and generate a COP packet that includes the encoded audio frame irrespective of the audio format. The COP packet is generated with a packet length proportional to a PCM length of the PCM frame, allowing the COP packetizing circuit to transmit the COP packet over an isochronous PCM interface with a lower bit rate than the PCM frame to reduce power. The COP packetizing circuit provides a mobile computing device with a single packetizing scheme that supports multiple audio formats, and allows for reducing power through bit rate scaling.

Abstract: A hearing aid system and method is disclosed. Disclosed embodiments provide for low latency enhanced audio using a hearing aid earpiece and an auxiliary processing unit wirelessly connected to the earpiece. These and other embodiments are disclosed herein.

Abstract: Subject of the invention is an apparatus described by a schematic block diagram for processing an audio signal to obtain a processed audio signal. The apparatus includes a phase calculator for calculating phase values for spectral values of a sequence of frequency-domain frames representing overlapping frames of the audio signal. Moreover, the phase calculator is configured to calculate the phase values based on information on a target time-domain envelope related to the processed audio signal, so that the processed audio signal has at least in an approximation the target time-domain envelope and a spectral envelope determined by the sequence of frequency-domain frames.

Abstract: An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

Abstract: A time scaler for providing a time scaled version of an input audio signal is configured to compute or estimate a quality of a time scaled version of the input audio signal obtainable by a time scaling of the input audio signal. The time scaler is configured to perform the time scaling of the input audio signal in dependence on the computation or estimation of the quality of the time scaled version of the input audio signal obtainable by the time scaling. An audio decoder has such a time scaler.

Abstract: A processor is configured to transition in and out of a low-power state at a first rate and to operate in a first mode or a second mode. In a particular method, the processor while coupled to a coder/decoder (CODEC) retrieves audio feature data from a buffer after transitioning out of the low-power state. The CODEC is configured to operate at a second rate in the first mode and at a third rate in the second mode, the second rate and the third rate each greater than the first rate. The audio feature data indicates features of audio data received during the low-power state of the processor. A ratio of CODEC activity to processor activity in the second mode is less than the ratio in the first mode.

Abstract: A method of combining data streams from fixed audio-visual sensors with data streams from personal mobile devices including, forming a communication link with at least one of one or more personal mobile devices; receiving at least one of an audio data stream and/or a video data stream from the at least one of the one or more personal mobile devices; determining the quality of the at least one of the audio data stream and/or the video data stream, wherein the audio data stream and/or the video data stream having a quality above a threshold quality is retained; and combining the retained audio data stream and/or the video data stream with the data streams from the fixed audio-visual sensors.

Abstract: An apparatus for enhancing a speech signal includes a spectral analysis circuit that generates time-frequency spectra of signals from a microphone array, a signal separation filter, and a hardware processor. The hardware processor identifies a key phrase in the signals and aligns a template of the key phrase to time-frequency spectra of a microphone in the array. A mask is generated from the aligned template and applied to the time-frequency spectra. The masked spectra are used to determine acoustic parameters that, in turn, are used to generate coefficients for configuring the source separation filter which is then used to process the spectra to generate an enhanced speech signal. The apparatus maybe used as a front end for a speech recognition engine, for example.

Abstract: An approach is provided for creating a digital representation of an analog sound. The approach retrieves a number of digital sound data streams with each of the digital sound data streams corresponding to an orientation angle of the digital sound data streams with respect to one another. The digital representation of the analog sound is generated by processing the digital sound data streams and their corresponding orientation angles.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: It is proposed a method for controlling the transmission of a sampled voice signal over a Bluetooth® radio link. The method comprises the steps of: a) monitoring the sampled voice signal for detecting whether the sampled voice signal comprises a number of consecutive silence samples; and b) halting the transmission of the sampled voice signal over the Bluetooth® radio link after having transmitted said consecutive silence samples.

Abstract: The present invention relates to a method and device for quantizing voice signals in a band-selective manner. A voice decoding method may include inversely quantizing voice parameter information produced from a selectively quantized voice band and performing inverse transform on the basis of the inversely quantized voice parameter information. Thus, according to the present invention, coding/decoding efficiency in voice coding/decoding may be increased by selectively coding/decoding important information.

Abstract: A method for quantizing phase information on an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a prototype pitch period signal based on the speech signal and transforming the prototype pitch period signal into a first frequency-domain signal. The method additionally includes mapping the first frequency-domain signal into a plurality of subbands. The method also includes determining a global alignment based on the first frequency-domain signal and quantizing the global alignment utilizing scalar quantization to obtain a quantized global alignment. The method additionally includes determining a plurality of band alignments corresponding to the plurality of subbands. The method also includes quantizing the plurality of band alignments utilizing vector quantization to obtain a quantized plurality of band alignments. The method further includes transmitting the quantized global alignment and the quantized plurality of band alignments.

Abstract: The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MP EG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table, wherein the matrix transformation coefficients below -60 Decibels are neglected.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

Abstract: A computer-implemented method and apparatus are disclosed for decoding an encoded data signal. In one embodiment, the method includes accessing, in a memory, a set of signal elements. The encoded data signal is received at a computing device. The signal includes signal fragments each having a projection value and an index value. The projection value has been calculated as a function of at least one signal element of the set of signal elements and at least a portion of the data signal. The index value associates its respective signal fragment with the at least one signal element used to calculate the projection value. The computing device determines amplitude values based on the projection values in the signal fragments. The decoded signal is determined using the amplitude values and the signal elements associated with the at least some of the signal fragments.

Abstract: Disclosed are a method for implementing a video call with a Bluetooth-based headset and a video communication terminal. The method includes: looking up a Bluetooth headset; starting a VTCALL unit of the AP module and an AG unit; sending a VTCALL command and an AG command to the audio module; switching an audio channel to a BTVTCALL channel; starting a recording and playing mode of the video communication terminal, collecting uplink and downlink audio data, and transmitting the uplink and the downlink audio data to the Bluetooth module; transmitting the uplink and the downlink audio data to the Bluetooth headset; executing the data exchange with the Bluetooth headset for implementing a Bluetooth video call. The present invention implements the Bluetooth video call by modifying software, thereby using the current hardware resources and decreasing the cost.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: Decomposition of a multi-source signal using a basis function inventory and a sparse recovery technique is disclosed.

Abstract: A method and an apparatus for encoding and decoding audio signals using adaptive sinusoidal coding are provided. The audio signal encoding method includes the steps of dividing a synthesized audio signal into a plurality of sub-bands, calculating the energy of each sub-band, selecting a predetermined number of sub-bands having a relatively large amount of energy from the sub-bands, and performing sinusoidal coding with regard to the selected sub-bands. Application of sinusoidal coding based on consideration of the amount of energy of each sub-band of the synthesized signal improves the quality of the synthesized signal more efficiently.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: An uplink or downlink audio processor contains a multi band compressor that receives an input, uplink or downlink, audio signal. The multi-band compressor has a band splitter that splits the input audio signal into a number of different band signals. Each band signal is input to a respective compressor block, which is independently programmable so that its audio frequency response (a) differs from a linear response in at least two non-overlapping windows of its input signal, and (b) differs from the frequency response of another one of the compressor blocks. Other embodiments are also described and claimed.

Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: The invention relates to a technique of operating a call control node controlling at least one section of a call path. The call path includes between two opposite edge nodes a multi-section harmonization path along which codec selection is to be harmonized. A method embodiment of the technique, wherein the call control node is a transfer node in the harmonization path between the edge nodes, comprises the steps of determining if the call control node is a transfer node of the harmonization path; determining if a codec used for the at least one section controlled by the call control node fulfills a predefined harmonization criterion; and providing, in case the used codec does not fulfill the harmonization criterion, a harmonization trigger indication to at least one of the edge nodes of the harmonization path for initiating harmonization.

Abstract: One or more methods and systems of detecting or identifying one or more types of algorithms used in the encoding of a voice or speech waveform is presented. The system and method may be used as a testing tool to identify whether a voice data stream is encoded using a linear G.711, ?-law G.711, or A-law G.711 algorithm. The system and method are applied to a voice data stream to ensure that a codec with the appropriate algorithm is used to reproduce an audio waveform.

Abstract: A system and method for redundant transmission is provided. In one embodiment, an input signal S is encoded as a list of fragments. Each fragment includes an index value and a projection value. The index points to an entry in a dictionary of signal elements. A repetition factor is assigned to each fragment based on its importance. After a fragment is added, a reconstructed signal is generated by decoding the list of fragments. Encoding terminates once the reconstructed signal is sufficiently close to the original signal S.

Abstract: In a sound processing device, a modulation spectrum specifier specifies a modulation spectrum of an input sound for each of a plurality of unit intervals. An index calculator calculates an index value corresponding to a magnitude of components of modulation frequencies belonging to a predetermined range of the modulation spectrum. A determinator determines whether the input sound of each of the unit intervals is a vocal sound or a non-vocal sound based on the index value.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes.

Abstract: In an encoding process, a CPU transforms an audio signal from the real-time domain to the frequency domain, and transforms the signal into spectra consisting of MDCT coefficients. The CPU separates the audio signal into several frequency bands, and performs bit shifting in each band such that the MDCT coefficients can be expressed with pre-configured numbers of bits. The CPU re-quantizes the MDCT coefficients at a precision differing for each band, and transmits the values acquired thereby and shift bit numbers as encoded data. Meanwhile, in a decoding process, a CPU receives encoded data and inverse re-quantizes and inverse bit shifts the data, thereby restoring the MDCT coefficients. Furthermore, the CPU transforms the data from frequency domain to the real-time domain by using the inverse MDCT, and restores and outputs the audio signal.

Abstract: A method for discriminating a telephony content signal into a first category or a second category is described. The method comprises a filtering procedure for obtaining from the telephony content signal a band signal set comprising one or more band signals, each band signal being associated with a respective frequency band at least one of said band signals being a sub-band signal (n) associated with a sub-band of an overall frequency band of the telephony content signal. Furthermore a determination procedure is provided for determining a band signal variation value (LLn) and a band signal strength value (TLn) for each band signal (n) of said band signal set. Finally, a discrimination procedure discriminates whether the telephony content signal is of the first category or of the second category.

Abstract: An encoding system includes a sampling unit, a computing unit, a comparing unit, a quantifying unit, and an encoding unit. The sampling unit obtains first sample data of a current sampling point and second sample data of a previous sampling point. The computing unit computes a data difference between the first sample data and the second sample data. The data difference includes a numeral and a sign. The comparing unit determines whether the data difference is more than or equal to 0 and outputs a determining result. The quantifying unit quantifies the numeral of the data difference. The encoding unit encodes the numeral of the data difference with or without the sign according to the determining result.

Abstract: The embodiments of a transcoding method, a transcoding device, and a communication apparatus are provided. The embodiment of a method includes: receiving a bit stream input from a sending end; determining an attribute of discontinuous transmission (DTX) used by a receiving end and a frame type of the input bit stream; and transcoding the input bit stream in a corresponding processing manner according to a determination result. Thereby, a corresponding transcoding operation is performed on the input bit stream according to the attribute of DTX used by the receiving end and the frame type of the input bit stream. In such a manner, input bit streams of various types can be processed, and the input bit streams can be correspondingly transcoded according to the requirements of the receiving end. Therefore, the average computational complexity and peak computational complexity can be effectively decreased without decreasing the quality of the synthesized speech.

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.