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48 results about "Advanced Audio Coding" patented technology
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Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at the same bit rate.
Reducing scale factor transmission cost for MPEG-2 advanced audio coding (AAC) using a lattice based post processing technique
InactiveUS7272566B2Bit cost can be reducedIncreased scale factorGain controlSpeech analysisAudio frequencyAudio signal
A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.
Owner:DOLBY LAB LICENSING CORP
Method and apparatus for coding audio signal
InactiveUS20060253276A1Reduce calculationImprove efficiencyMicrometersSpeech analysisAudio frequencyAudio signal
An audio coding method and apparatus capable of improving efficiency of a MPEG-4 AAC (Moving Picture Expert Group-4 Advanced Audio Coding) process are disclosed. The audio coding method and apparatus reduce the number of calculations of an audio coding algorithm to improve efficiency of an audio coding process. Specifically, the audio coding method and apparatus reduce the number of calculations required for a Psychoacoustic model process of the MPEG-4 AAC algorithm capable of coding an audio signal.
Owner:LG ELECTRONICS INC +1
Audio processing method in mobile digital television recording
ActiveCN102254560ASmooth playbackBit error noTelevision system detailsPulse modulation television signal transmissionComputer hardwareData transport
The invention discloses an audio processing method in mobile digital television recording. The method comprises the following steps of: packaging each advanced audio coding + (AAC+) code stream naked data frame which is received from the air into a frame in an audio data transport stream (ADTS) format; analyzing an ADTS frame head to find the original AAC+ code stream naked data of the frame; decoding the AAC+ code stream naked data, removing spectral band replication (SBR) information from the AAC+ code stream naked data in an audio decoding, recording and buffering area, and converting the AAC+ code stream naked data in the audio decoding, recording and buffering area into AAC naked data; and performing a subsequent decoding process by using an AAC+ decoder, and transmitting the AAC naked data to a recording module to record video and audio files. By the method, the mobile digital television recording also can be performed even if the signal is poor, and a recorded video file can benormally played on a computer by mainstream video play software, and severe abnormal phenomena such as harsh noise, silence, player breakdown, system crash and the like are avoided.
Owner:ANYKA (GUANGZHOU) MICROELECTRONICS TECH CO LTD
Method and apparatus for svc video and aac audio synchronization using npt
InactiveUS20100329360A1Reduce in quantityReduce the amount of controlPulse modulation television signal transmissionPicture reproducers using cathode ray tubesVideo encodingReal-time Transport Protocol
A method of supporting synchronization of Scalable Video Coding (SVC) information and Advanced Audio Coding (AAC) information using a Normal Play Time (NPT), the method including: receiving video information using a decoding apparatus; receiving audio information using the decoding apparatus; calculating the NPT of the video information using a Real-time Transport Protocol (RTP) time stamp included in the received video information; calculating the NPT of the audio information using the RTP time stamp included in the received audio information; comparing the NPT of the video information and the NPT of the audio information to calculate a difference value; determining whether the calculated difference value is included in a specific synchronization region; and outputting the audio information and the video information when the calculated difference value is determined to be included in the specific synchronization region.
Owner:ELECTRONICS & TELECOMM RES INST
Compressed data multiple description transmission and resolution conversion system
InactiveUS20060257036A1Color television with pulse code modulationColor television with bandwidth reductionHigh definition tvInternational standard
A system provides lossless split and merge processes of integer discrete cosine transform (DCT) transformed data such that the discrete cosine transform of one data block may be split into two half length DCT odd and even blocks for merging, with split and merge processes being lossless and are generated in the discrete cosine transformed domain. After splitting, the redundancy existing between the two integer discrete cosine transformed half data blocks allows one to approximately reconstruct the original data block in case one of the discrete cosine transformed half data block is lost during transmission. The system may be used with existing JPEG and MPEG compressors and decompressors because both use the discrete cosine transform for image and video compression and decompression, may be used as a resolution conversion device for transcribing from digital high-definition TV to analog low-definition TV, and may be used for lossless splitting and merging type-IV discrete cosine transformed data for audio compression and decompression in the international standard MPEG-4 Advanced Audio Coding (AAC), such as AC-3 or MP3.
Owner:THE AEROSPACE CORPORATION
Computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC audio decoding algorithm on programmable processors
InactiveUS20070024472A1Reduce chip sizeSmall small amount of powerSpeech analysisCode conversionAdvanced Audio CodingComputer engineering
The present invention relates to the computing method and Huffman computing circuits for improving the correctness and efficiency of the nonlinear inverse quantization when the MPEG-2 AAC (Advanced Audio Coding) or MPEG-4 AAC algorithm which is used as an audio compression algorithm in multi-channel high-quality audio systems is implemented on programmable processors. In accordance with the present invention, while the architecture of the existing digital signal processor is reused, the performance can be improved by means of the addition of Huffman decoder and bit processing architecture. Accordingly, to design and change the programmable processor can be facilitated.
Owner:AJOU UNIV IND ACADEMIC COOP FOUND +1
Effective deployment of temporal noise shaping (TNS) filters
InactiveUS7099830B1Efficient deploymentSpeech analysisTransmission noise suppressionTime structureFrequency spectrum
In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The AAC standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Effective deployment of temporal noise shaping (TNS) filters
In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The AAC standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Audio encoder and audio encoding method
InactiveCN101350199AMeet the requirements of the specificationSpeech analysisTime domainMasking threshold
The invention provides an audio encoder and an audio coding method, which are used for advanced audio coding, wherein the encoder comprises a spectrum processing module, a quantification and bit distribution module and a bit encapsulated module which are orderly connected. The invention further comprises a compound modified forward discrete cosine transform module, a real part operation module and a psychoacoustic model module, wherein the compound modified forward discrete cosine transform module is used to execute compound modified forward discrete cosine transform for audio time domain data which is received, thereby producing compound modified forward discrete cosine transform frequency domain spectrum data. The real part operation module is used to execute a real part operation for the compound modified forward discrete cosine transform frequency domain spectrum data which is output by the compound modified forward discrete cosine transform module, thereby obtaining modified forward discrete cosine transform frequency domain spectrum data and sending to the spectrum processing module. The psychoacoustic model module is used to analyze perceptual characteristics of audio singles through the compound modified forward discrete cosine transform frequency domain spectrum data, thereby obtaining a masking threshold of the audio singles and sending to the quantification and bit distribution module. The invention reduces the computational complexity, and reduces the demand quantity of memory.
Owner:VIMICRO CORP
Computing circuits and method for running an MPEG-2 AAC or MPEG-4 AAC audio decoding algorithm on programmable processors
InactiveUS7333036B2Easy to operateSmall small amountSpeech analysisCode conversionHuffman decodingInverse quantization
Owner:AJOU UNIV IND ACADEMIC COOP FOUND +1
Compressed data multiple description transmission and resolution conversion system
InactiveUS7720299B2Color television with pulse code modulationColor television with bandwidth reductionHigh definition tvOriginal data
A system provides lossless split and merge processes of integer discrete cosine transform (DCT) transformed data such that the discrete cosine transform of one data block may be split into two half length DCT odd and even blocks for merging, with split and merge processes being lossless and are generated in the discrete cosine transformed domain. After splitting, the redundancy existing between the two integer discrete cosine transformed half data blocks allows one to approximately reconstruct the original data block in case one of the discrete cosine transformed half data block is lost during transmission. The system may be used with existing JPEG and MPEG compressors and decompressors because both use the discrete cosine transform for image and video compression and decompression, may be used as a resolution conversion device for transcribing from digital high-definition TV to analog low-definition TV, and may be used for lossless splitting and merging type-IV discrete cosine transformed data for audio compression and decompression in the international standard MPEG-4 Advanced Audio Coding (AAC), such as AC-3 or MP3.
Owner:THE AEROSPACE CORPORATION
Method and apparatus for coding audio signal
An audio coding method and apparatus capable of improving efficiency of a MPEG-4 AAC (Moving Picture Expert Group-4 Advanced Audio Coding) process are disclosed. The audio coding method and apparatus reduce the number of calculations of an audio coding algorithm to improve efficiency of an audio coding process. Specifically, the audio coding method and apparatus reduce the number of calculations required for a Psychoacoustic model process of the MPEG-4 AAC algorithm capable of coding an audio signal.
Owner:LG ELECTRONICS INC +1
Real-time container conversion realization method and device of LATM (low-overhead Moving Picture Experts Group-4 audio trans-port multiplex) AAC (advanced audio coding) audio stream.
ActiveCN108122558AImplement automatic identification processing mechanismAdd supportSpeech analysisTransmissionSystem integrationData stream
The invention belongs to the field of audio conversion and discloses a real-time container conversion realization method and a device of an LATM (low-overhead Moving Picture Experts Group-4 audio trans-port multiplex) AAC (advanced audio coding) audio stream. The method comprises the following steps of: step 1, receiving and processing an audio data stream and reading audio frame data, step 2, judging the head of the audio frame data is of an LATM AAC data format and going to step 3 if so, directly going to step 5 if not, step 3, analyzing the audio frame data and acquiring a core parameter and audio load data of an LATM AAC audio frame, step 4, forming an ADTS (audio data transport stream) AAC audio frame, and step 5, outputting the audio frame. The method and the device effectively achieves judgment of an LATM AAC audio in a transport code stream, container resolution and container conversion processing to ADTS AAC; an audio decoding library can support the LATM AAC audio stream well; and the system integration cost is lowered.
Owner:SHENZHEN STATE MICRO TECH CO LTD
Coding parameter statistical feature-based AAC sound recording document source identification method
InactiveCN105845132AAccurate identificationImprove recognition accuracySubstation equipmentSpeech recognitionMobile phoneAdvanced Audio Coding
The invention discloses a coding parameter statistical feature-based AAC (advanced audio coding) sound recording document source identification method. According to the design thought of the invention, the use features and statistical features of a plurality of coding parameters of AAC sound recording documents of smart phones are analyzed; the tendencies and characteristics of using the coding parameters when the mobile phones generate the AAC sound recording documents are found out; and features for distinguishing phone models are constructed; and therefore, accurate identification of the sources of the AAC sound recording documents can be realized. The AAC sound recording document source identification method has the advantages of high recognition accuracy, convenience in operation and the like.
Owner:NINGBO UNIV
Universal steganalysis method and system of audio based on spectrograms and deep residual network
The invention discloses a universal steganalysis method and system of audio based on spectrograms and deep residual network. Targeting at the current situation that existing steganographic algorithmsbased on audio compression standards perform steganography by correcting different audio compression parameters while no universal steganalysis algorithm is present, the method of the invention includes extracting spectrogram features of a recompressed original audio signal by comprehensively considering common MDCT (modified discrete cosine transform) features in AAC (advanced audio coding) and other compression coding standards, mining inherent distribution features of the audio signal via the deep residual network S-ResNet, and extracting classification features to construct a universal audio steganalysis unit. The method and system of the invention have the advantages that the method and system are not limited to a single coding standard and parameter domain and the method and system have good universality and good steganalysis test performance.
Owner:WUHAN UNIV
Effective deployment of temporal noise shaping (TNS) filters
The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a“short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Audio data playing method and device
InactiveCN106303754AReduce transmission packet drop rateSolve the problem that M3U8 files in AAC format cannot be playedSelective content distributionComputer hardwareData segment
The invention discloses an audio data playing method and a device and solves a problem that an Android system player can not play M3U8 files in an AAC format. The method comprises steps that a request for downloading the audio data is sent to a server; multiple M3U8 fragment files sent by the server are received; an index address of a corresponding high-grade audio-coding-format AAC audio is recorded in the multiple M3U8 fragment files; an index address of an AAC audio data segment corresponding to each M3U8 fragment file is acquired through analyzing the multiple received M3U8 fragment files; according to the index addresses of the multiple M3U8 fragment files acquired through analysis, an AAC audio data segment stored in each index address is acquired; the acquired multiple AAC audio data segments are played in a data stream mode. The method is advantaged in that a client especially an Android system client is enabled to directly play audio streams, and strong compatibility is realized.
Owner:TVMINING BEIJING MEDIA TECH
Noise detection for audio encoding by mean and variance energy ratio
The techniques described are utilized for detection of noise and noise-like segments in audio coding. The techniques can include performing a prediction gain calculation, an energy compaction calculation, and a mean and variation energy calculation. Signal adaptive noise decisions can be made both in time and frequency dimensions. The techniques can be embodied as part of an AAC (advanced audio coding) encoder to detect noise and noise-like spectral bands. This detected information is transmitted in a bitstream using a signaling method defined for a perceptual noise substitution (PNS) encoding tool of the AAC encoder.
Owner:NOKIA TECH OY
Method and Apparatus of Low-Complexity Psychoacoustic Model Applicable for Advanced Audio Coding Encoders
InactiveUS20090063137A1Maintain qualityImprove efficiencySpeech analysisComputation complexityAcoustic model
A method and an apparatus of a low-complexity psychoacoustic model applicable for advanced audio coding encoders use a modified discrete cosine transform based (MDCT-based) psychoacoustic model and a simplified look-up table to compute the MDCT-based psychoacoustic model by a logarithm based logarithmic method to simplify the computational complexity, and then computing a quantization loop (Q loop) by the logarithm based logarithmic method to further reduce the computational quantity of the MDCT-based psychoacoustic model, so as to achieve the real-time playback effect by a very low operating frequency.
Owner:NAT CENT UNIV
Memory optimization method for realizing advanced audio coding algorithm on digital signal processor (DSP)
InactiveCN101968771AReduce usageReduce incidenceMemory adressing/allocation/relocationSpeech analysisDigital signal processingStatic memory allocation
The invention provides a memory optimization method for realizing an advanced audio coding algorithm on a digital signal processor (DSP), which comprises the following steps of: transporting data and code stream by using a direct memory access mechanism; performing external static memory allocation on a frequent coding structure in the coding process and calling the structure in a coding function by using a pointer to reduce memory fragments; and in the memory allocation process of a stack, determining the using depth of the stack first and then allocating a memory to the stack in a DSP configuration file by using the using depth of the stack as a basis. The method not only saves the system memories of the DSP but also improves data transmission efficiency and memory using efficiency, so that the advanced audio coding algorithm is realized under the condition of finite system resources of the DSP.
Owner:BEIHANG UNIV
MDCT quantization coefficient small value region-based advanced audio coding (AAC) audio steganography and extraction method
ActiveCN104867496AReduce computational complexityIncrease imperceptibilitySpeech analysisSteganalysisComputation complexity
The invention discloses an MDCT quantization coefficient small value region-based advanced audio coding (AAC) audio steganography and extraction method. The method includes the following steps that: 1, a mapping table containing index pairs and code length is established according to an AAC audio codebook; steganography rules are established according to the mapping table; and 3, insertion and extraction of secret information can be realized through utilizing the mapping table and the steganography rules. With the method of the invention adopted, computational complexity in steganography and extraction processes can be decreased, and imperceptibility and anti-steganalysis of steganography can be improved, and therefore, information can be hidden in and extracted from AAC audios safely and efficiently.
Owner:合肥庐阳科技创新集团有限公司
Method and device for processing audio data
InactiveCN104795073AImprove coding efficiencyReduce data volumeSpeech analysisPulse-code modulationData format
The invention discloses a method and a device for processing audio data. The method includes acquiring the audio data in pulse code modulation (PCM) data formats; framing the acquired PCM audio data to obtain PCM data frames; calling at least two concurrent coding threads and distributing the PCM data frames to the concurrent coding threads according to preset strategies; carrying out AAC-ELD (advanced audio coding-enhanced low display) coding on the PCM data frames by the aid of the concurrent coding threads to obtain coded AAC-ELD data frames; transmitting the AAC-ELD data frames. The method and the device have the advantages that the audio data coding efficiency can be improved, and accordingly the purpose of coding the audio data in real time can be achieved.
Owner:WUXI TVMINING MEDIA SCI & TECH
AAC audio double compression detection method based on QMDCT coefficient
ActiveCN108877816AImprove detection accuracyReduce dimensionalitySpeech analysisFeature vectorComputation complexity
The present invention discloses an AAC (Advanced Audio Coding) audio double compression detection method based on a QMDCT (Quantized Modified Discrete Cosine Transform) coefficient. The method comprises the steps of: obtaining single compression AAC audio and a dual compression AAC audio with different bit rate; removing sampling points to obtain single compression removal sampling point AAC audios and dual compression removal sampling point AAC audios; obtaining a corresponding feature vector according to QMDCT coefficient distribution histograms of the single compression AAC audio and the corresponding single compression removal sampling point AAC audios; and obtaining a corresponding feature vector according to QMDCT coefficient distribution histograms of the dual compression AAC audioand the corresponding dual compression removal sampling point AAC audios; training a LIBSVM classifier according to the feature vectors of the single compression AAC audios and the dual compression AAC audios; and when test is performing, inputting the bit rate of the AAC audio to be subjected to dual compression detection to a trained corresponding LIBSVM classifier to obtain a detection result.The AAC audio double compression detection method based on the QMDCT coefficient achieves effective detection of the AAC audios of the dual compression from low code rate to high code rate and dual compression with the same code rate, and is high in detection accuracy, low in computing complexity and high in robustness.
Owner:NINGBO UNIV
Improved advanced audio encoding/decoding method and system of wireless Bluetooth system
The invention provides the improved advanced audio coding / decoding method of a wireless Bluetooth system. The method comprises the following steps that a main audio device carries out AAC partial decoding on the received audio signal of AAC coding; noiseless decoding, inverse quantization processing, and combined stereo decoding are performed on the received audio signal of AAC coding to obtain the audio signal of a decoded intermediate state; one path of the audio signal is selected from the audio signal of the decoded intermediate state obtained through AAC partial decoding to carry out AACpartial coding; quantitative processing and noiseless coding are performed on the one path of the selected intermediate state of audio signal to obtain the audio signal of AAC partial coding; and themain audio device forwards the audio signal of the AAC partial coding to a slave audio device through Bluetooth. The operand of the main audio device can be effectively reduced and the time delay of awhole system is decreased.
Owner:BESTECHNIC SHANGHAI CO LTD
Scrambled/descrambled-data-scattering-based video scrambling system
InactiveCN102238418ASolve a practical problemLow costSelective content distributionMultiplexingConditional access systems
Owner:昆明亿尚科技有限公司
FAAD2 MAIN mode-based multipath audio real-time decoding software design method
The invention provides an FAAD MAIN mode-based multipath audio real-time decoding software design method. The software design method mainly comprises a multipath audio receiving mechanism module, multipath filter bank preserved buffer zones and a multipath audio transmitting mechanism module, wherein the multipath audio receiving mechanism module comprises multipath receiving transmission buffer zones, and each path of receiving transmission buffer zone can store 2 frames of advanced audio coding (AAC) code streams to prevent data overflow and ensure that an AAC decoder correctly receives multipath audio data; each path of filter bank preserved buffer zone stores pulse code modulation (PCM) data after the last frame of decoding data is subjected to inverse modified discrete cosine transform (IMDCT), and performs time domain superposition by utilizing PCM data in the filter bank preserved buffer zone of the current link and the PCM data after the current decoding data is subjected to IMDCT to obtain output audio data; and the multipath audio transmitting mechanism module comprises multipath transmitting transmission buffer zones, and each path of transmitting transmission buffer zone stores 1 frame of output audio data so as to ensure that the AAC decoder correctly transmits multipath output audio data.
Owner:BEIHANG UNIV
Audio processing method in mobile digital television recording
ActiveCN102254560BSmooth playbackBit error noTelevision system detailsSpeech analysisComputer hardwareData transport
The invention discloses an audio processing method in mobile digital television recording. The method comprises the following steps of: packaging each advanced audio coding + (AAC+) code stream naked data frame which is received from the air into a frame in an audio data transport stream (ADTS) format; analyzing an ADTS frame head to find the original AAC+ code stream naked data of the frame; decoding the AAC+ code stream naked data, removing spectral band replication (SBR) information from the AAC+ code stream naked data in an audio decoding, recording and buffering area, and converting the AAC+ code stream naked data in the audio decoding, recording and buffering area into AAC naked data; and performing a subsequent decoding process by using an AAC+ decoder, and transmitting the AAC naked data to a recording module to record video and audio files. By the method, the mobile digital television recording also can be performed even if the signal is poor, and a recorded video file can benormally played on a computer by mainstream video play software, and severe abnormal phenomena such as harsh noise, silence, player breakdown, system crash and the like are avoided.
Owner:ANYKA (GUANGZHOU) MICROELECTRONICS TECH CO LTD
FAAC and FAAD2-based single track constant bit rate audio realtime coding and decoding error correcting method
InactiveCN101968962ADoes not destroy frame structureTransmission control mechanisms are not affectedError preventionSpeech analysisHamming codeError checking
The invention provides a free advanced audio coder(FAAC) and AAC audio decoder (FAAD2)-based single track constant bit rate audio realtime coding and decoding error correcting method, which mainly comprises that: a coder performs Hamming coding of fixed and variable frame headers, the Hamming coding of ID-SCE, SCETAG, a global scale factor and ID-END in a group, and Hamming coding of side information, packs a data area according to the size of a padding data area, performs Hamming coding of each data packet, separates check bit data from Hamming codes and stores the check bit data in the padding data area in turn; and a decoder acquires check bit data from the packing data area and performs the Hamming code error checking and correcting of corresponding data bits. On the premise of ensuring the structural integrity of an advanced audio coding (AAC) frame, the method has the capability of correcting errors of AAC data frames.
Owner:BEIHANG UNIV
Voice quality inspection analysis method, device, equipment and medium
PendingCN114218427AImprove accuracyImprove service qualityCustomer relationshipSpeech analysisEngineeringClient data
The invention relates to the technical field of voice processing, and discloses a voice quality inspection analysis method, device, equipment and medium, and the method comprises the steps: predicting a customer identifier through obtaining reservation customer data and a business service list, and building the connection of a server through a Native connection method, the method comprises the following steps: receiving an audio stream file from a server side, sending reservation client data to the server side, adding the reservation client data into a client service list, when the audio stream file from the server side is received, performing audio coding conversion by using an advanced audio coding algorithm to obtain an audio file, and instructing the server side to notify a chest card to clean a space; performing audio quality inspection on the reserved customer data and the audio file through a quality inspection detection model to obtain a quality inspection result; and inputting the quality inspection result and each historical quality inspection result into a quality inspection clustering model, and carrying out graph clustering analysis to obtain a quality inspection analysis result. Therefore, the accuracy of the quality inspection result is improved, and the quality inspection analysis result of insufficient business items is automatically analyzed.
Owner:PING AN BANK CO LTD
Audio coding and decoding system
InactiveCN104183243AReduce resource consumptionControl power consumptionSpeech analysisResource consumptionAudio frequency
The invention brings forward an audio coding and decoding system. An advanced audio coding (AAC) and decoding algorithm is adopted by the audio coding and decoding system at the same audio coding and decoding unit (DSP). Meanwhile, high-fidelity audio coding and decoding are realized so that resource consumption is substantially reduced, and product power consumption, volume and weight are well controlled.
Owner:SHANGHAI SPACEFLIGHT INST OF TT&C & TELECOMM
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