Integration of high frequency reconstruction techniques with reduced post-processing delay

Abstract

This invention discloses a method for decoding an encoded audio bitstream. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded low-frequency band audio signal. The method further includes extracting high-frequency reconstruction post-processing data and using an analysis filter bank to filter the decoded low-frequency band audio signal to generate a filtered low-frequency band audio signal. The method also includes extracting a flag indicating whether to perform a spectral shift or harmonic transpose on the audio data and regenerating a high-frequency band portion of the audio signal based on the flag using the filtered low-frequency band audio signal and the high-frequency reconstruction post-processing data. The high-frequency regeneration is performed with a delay of 3010 samples per audio channel as part of a post-processing operation.

Description

Technical Field

[0001] The embodiments relate to audio signal processing, and more specifically, to encoding, decoding, or transcoding an audio bitstream having control data to specify a basic form or an enhanced form of high-frequency reconstruction ("HFR") of the audio data. Prior Technology

[0002] A typical audio bitstream comprises audio data (e.g., coded audio data) indicating one or more channels of audio content, and at least one characteristic indicating the audio data or audio content, both of which are then present. One well-known format used to generate a coded audio bitstream is the MPEG-4 Advanced Audio Coding (AAC) format described in the MPEG standard ISO / IEC 14496-3:2009. In the MPEG-4 standard, AAC stands for "Advanced Audio Coding," and HE-AAC stands for "High Efficiency Advanced Audio Coding."

[0003] The MPEG-4 AAC standard defines several audio profiles that determine which objects and encoding tools are present in a compatible encoder or decoder. These three audio profiles are (1) the AAC profile, (2) the HE-AAC profile, and (3) the HE-AAC v2 profile. The AAC profile includes the AAC Low Complexity (or "AAC-LC") object type. The AAC-LC object is the counterpart to the MPEG-2 AAC Low Complexity profile, with some adjustments, and does not include either the Spectral Band Replication ("SBR") object type or the Parametric Stereo ("PS") object type. The HE-AAC profile is a superset of the AAC profile and also includes the SBR object type. The HE-AAC v2 profile is a superset of the HE-AAC profile and also includes the PS object type.

[0004] The SBR object type contains a spectrum band copying tool, which is an important high-frequency reconstruction ("HFR") encoding tool that can significantly improve the compression efficiency of perceptual audio codecs. SBR reconstructs the high-frequency components of an audio signal at the receiver side (e.g., in the decoder). Therefore, the encoder only needs to encode and transmit the low-frequency components to allow for much higher audio quality at low data rates. SBR is based on a signal with limited available bandwidth obtained from an autoencoder and control data that has been previously truncated to reduce the data rate, resulting in a harmonic sequence. The ratio between tonal components and quasi-noise components is maintained through adaptive inverse filtering and, whereby, noise and sine waves are added as needed. In the MPEG-4 AAC standard, the SBR tool performs spectrum patching (also known as linear shifting or spectrum shifting), where several consecutive quadrature mirror filter (QMF) subbands are copied (or "patched") from a low-frequency portion of an audio signal to a high-frequency portion of that audio signal (which is generated in the decoder).

[0005] Spectral patching or linear shifting may not be suitable for certain audio types (such as music with relatively low crossover frequencies). Therefore, techniques for improving spectral band replication are needed. Summary of the Invention

[0006] A first-type embodiment relates to a method for decoding an encoded audio bitstream. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded low-frequency band audio signal. The method further includes extracting high-frequency reconstruction post-processing data and using an analysis filter bank to filter the decoded low-frequency band audio signal to generate a filtered low-frequency band audio signal. The method further includes extracting a flag indicating spectral shift or harmonic transpose of the audio data and regenerating a high-frequency band portion of the audio signal based on the flag using the filtered low-frequency band audio signal and the high-frequency reconstruction post-processing data. Finally, the method includes combining the filtered low-frequency band audio signal and the regenerated high-frequency band portion to form a broadband audio signal.

[0007] A second type of embodiment relates to an audio decoder for decoding an encoded audio bitstream. The decoder includes an input interface for receiving the encoded audio bitstream (wherein the encoded audio bitstream contains audio data representing a low-frequency band portion of an audio signal) and a core decoder for decoding the audio data to generate a decoded low-frequency band audio signal. The decoder also includes a demultiplexer for extracting high-frequency reconstruction post-processing data from the encoded audio bitstream (wherein the high-frequency reconstruction post-processing data contains operating parameters for a high-frequency reconstruction procedure that linearly shifts several consecutive sub-bands from a low-frequency band portion of the audio signal to a high-frequency band portion of the audio signal) and an analysis filter bank for filtering the decoded low-frequency band audio signal to generate a filtered low-frequency band audio signal. The decoder further includes a demultiplexer for extracting a flag (indicating linear shift or harmonic transpose of the audio data) from the encoded audio bitstream, and a high-frequency regenerator for regenerating a high-frequency portion of the audio signal using the filtered low-frequency audio signal and the high-frequency reconstructed post-processor data based on the flag. Finally, the decoder includes a synthesis filter bank for combining the filtered low-frequency audio signal and the regenerated high-frequency portion to form a broadband audio signal.

[0008] Other embodiments relate to encoding and transcoding audio bitstreams, which contain information indicating whether enhanced spectrum band replication (eSBR) processing has been performed. Simple Explanation of the Diagram

[0009] Figure 1 is a block diagram of one embodiment of a system that can be configured to perform the method of the present invention.

[0010] Figure 2 is a block diagram of an encoder, which is an embodiment of the audio processing unit of the present invention.

[0011] Figure 3 is a block diagram of a system that includes a decoder (which is an embodiment of the audio processing unit of the present invention) and, where appropriate, a post-processor coupled to the decoder.

[0012] Figure 4 is a block diagram of a decoder, which is an embodiment of the audio processing unit of the present invention.

[0013] Figure 5 is a block diagram of one type of decoder, which is another embodiment of the audio processing unit of the present invention.

[0014] Figure 6 is a block diagram of another embodiment of the audio processing unit of the present invention.

[0015] Figure 7 is a block diagram of an MPEG-4 AAC bitstream, which contains several segments into which it is divided. Implementation

[0016] [ ] Cross-reference to related applications This application claims priority to U.S. Provisional Patent Application No. 62 / 662,296, filed April 25, 2018, the entire contents of which are incorporated herein by reference. Symbols and terms

[0017] In this invention (which is included in the scope of the claims), the expression "perform an operation on a signal or data" (e.g., filter the signal or data, adjust the signal or data proportionally, transform the signal or data, or apply a gain to the signal or data) is used broadly to mean "perform an operation directly on the signal or data or on one of the processed forms of the signal or data" (e.g., perform an operation on one of the forms of the signal that has undergone preliminary filtering or preprocessing before the operation is performed on the signal).

[0018] In this invention (including within the scope of the claims), the term "audio processing unit" or "audio processor" is used broadly to refer to a system, apparatus, or device configured to process audio data. Examples of audio processing units include (but are not limited to) encoders, transcoders, decoders, codecs, preprocessing systems, post-processing systems, and bitstream processing systems (sometimes referred to as bitstream processing tools). Almost all consumer electronics products (such as mobile phones, televisions, laptops, and tablets) contain an audio processing unit or audio processor.

[0019] In this invention (included in the scope of the claims), the term "coupled" is used broadly to mean a direct or indirect connection. Thus, if a first device is coupled to a second device, the connection can be a direct connection or an indirect connection via other devices and connections. Furthermore, components integrated into or integrated with other components are also coupled to each other.

[0020] The MPEG-4 AAC standard anticipates that an encoded MPEG-4 AAC bitstream contains post-processing data that indicates, and / or controls, various types of High Frequency Reconstruction ("HFR") processing applied (if to be applied) by a decoder to decode the audio content of the bitstream, and / or indicates at least one characteristic or parameter of at least one HFR tool used to decode the audio content of the bitstream. In this document, we use the term "SBR post-processing data" to refer to this type of post-processing data used in conjunction with Spectral Band Replication ("SBR"), as described or mentioned in the MPEG-4 AAC standard. Those skilled in the art will understand that SBR is a form of HFR.

[0021] SBR is better suited for use in a dual-rate system, where the basic codec operates at half the original sampling rate, while the SBR operates at the original sampling rate. Despite having a higher sampling rate, the SBR encoder operates in parallel with the basic core codec. Although the SBR is primarily a post-processing step in the decoder, it extracts important parameters from the encoder to ensure the most accurate high-frequency reconstruction in the decoder. The encoder estimates the spectral envelope of the SBR range that best suits the time and frequency range / resolution of the current input signal segment. The spectral envelope is estimated using a complex QMF analysis and subsequent energy calculation. The time and frequency resolution of the spectral envelope can be chosen with a high degree of freedom to ensure the most suitable time and frequency resolution for a given input segment. Envelope estimation needs to consider that a transient state of the original source (primarily located in the high-frequency region, such as a high cap) will exist in small quantities in the high-frequency band generated by the SBR before envelope adjustment, since the high-frequency band in the decoder is based on a low-frequency band where transient states are much less pronounced than in the high-frequency band. Compared to general spectral envelope estimation used in other audio coding algorithms, this sample places different requirements on the time-frequency resolution of the spectral envelope data.

[0022] In addition to the spectral envelope, several additional parameters representing the spectral characteristics of the input signal in different time and frequency regions are extracted. Since the encoder inherently has access to the original signal and information about how the SBR unit in the decoder will generate the high-frequency band, given a specific set of control parameters, the system can handle situations where the low-frequency band constitutes a strong harmonic series and the high-frequency band to be regenerated mainly consists of random signal components, as well as situations where strong tone components exist in the original high-frequency band (which does not have corresponding objects in the low-frequency band upon which the high-frequency band region is based). Furthermore, the SBR encoder works closely with the basic core codec to evaluate which frequency range should be covered by the SBR at a given time. For stereo signals, the SBR data is efficiently encoded before transmission using entropy coding and the channel dependency of the control data.

[0023] Typically, the algorithm needs to be carefully tuned to extract control parameters based on the basic codec at a given bit rate and a given sampling rate. This is because a lower bit rate usually implies a larger SBR range than a higher bit rate, and different sampling rates correspond to different time resolutions of the SBR frame.

[0024] An SBR decoder typically comprises several distinct components. These include a bitstream decoding module, a high-frequency reconstruction (HFR) module, an additional high-frequency component module, and an envelope adjuster module. The system is based on either a complex-valued QMF filter bank (for high-quality SBR) or a real-valued QMF filter bank (for low-power SBR). Embodiments of this invention are applicable to both high-quality and low-power SBR. In the bitstream extraction module, control data is read from the bitstream and decoded. Before reading the envelope data from the bitstream, the time-frequency grid of the current frame is obtained. The basic core decoder decodes the audio signal of the current frame (albeit at a lower sampling rate) to generate time-domain audio samples. The resulting frame from the audio data is used by the HFR module for high-frequency reconstruction. Next, a QMF filter bank is used to analyze the decoded low-frequency band signal. Subsequently, high-frequency reconstruction and envelope adjustment are performed on the sub-band samples of the QMF filter bank. Based on given control parameters, high frequencies are reconstructed from the low-frequency band in a flexible manner. Furthermore, based on control data, high-frequency bands are reconstructed through adaptive filtering using a sub-band channel to ensure appropriate spectral characteristics for a given time / frequency region.

[0025] The top layer of an MPEG-4 AAC bitstream is a sequence of data blocks (“raw_data_block” elements), each of which is a data segment (referred to herein as a “block”) containing audio data (typically within a time interval of 1024 or 960 samples) and related information and / or other data. In this document, we use the term “block” to refer to a segment of an MPEG-4 AAC bitstream that includes audio data (and corresponding meta-data and, where applicable, other related data) (which determines or indicates one (but not more than one) “raw_data_block” element).

[0026] Each block of an MPEG-4 AAC bitstream can contain several syntax elements (each of which is also materialized as a data segment in the bitstream). The MPEG-4 AAC standard defines seven types of these syntax elements. Each syntax element is identified by a distinct value of the data element "id_syn_ele". Instances of syntax elements include a "single_channel_element()", a "channel_pair_element()", and a "fill_element()". A single-channel element is a region containing audio data for a single audio channel (a mono audio signal). A channel-pair element contains audio data for two audio channels (i.e., a stereo audio signal).

[0027] A padding element is an information region containing an identifier (such as the value of the element "id_syn_ele" mentioned above) and subsequent data (referred to as "padding data"). Padding elements have historically been used to adjust the instantaneous bit rate of a bit stream that will be transmitted through a constant-rate channel. A constant data rate can be achieved by adding an appropriate amount of padding data to each block.

[0028] According to embodiments of the present invention, the padding data may include one or more extended payloads that extend the data types (e.g., post-data) that can be transmitted in a bitstream. A decoder (which receives a bitstream having padding data containing a new data type) may be used by a receiving bitstream device (e.g., a decoder) to extend the functionality of that device, as appropriate. Therefore, those skilled in the art will understand that the padding element is a special type of data structure and is different from the data structure typically used for transmitting audio data (e.g., audio payload containing channel data).

[0029] In some embodiments of the present invention, the identifier used to identify a padding element may consist of a 3-bit unsigned integer, with the most significant bit ("uimsbf") transmitted first (which has a value of 0×6). Within a block, several instances of the same type of syntax element (e.g., several padding elements) may appear.

[0030] Another standard used for encoding audio bitstreams is the MPEG Unified Speech and Audio Coding (USAC) standard (ISO / IEC 23003-3:2012). The MPEG USAC standard describes the use of spectrum band copying processing (including the SBR processing described in the MPEG-4 AAC standard and other enhancements to spectrum band copying processing) to encode and decode audio content. This processing uses an extended and enhanced form of the spectrum band copying tool (sometimes referred to herein as the "enhanced SBR tool" or "eSBR tool") of the SBR toolkit described in the MPEG-4 AAC standard. Therefore, eSBR (as defined in the USAC standard) is an improvement on SBR (as defined in the MPEG-4 AAC standard).

[0031] In this document, we use the term "enhanced SBR processing" (or "eSBR processing") to refer to spectral band copying processing using at least one eSBR tool not described or mentioned in the MPEG-4 AAC standard (such as at least one eSBR tool described or mentioned in the MPEG USAC standard). Examples of such eSBR tools are harmonic transpose and QMF patching additional preprocessing or "pre-flattening".

[0032] An integer-order T harmonic transposer maps a sine wave with frequency ω to a sine wave with frequency Tω, while preserving the signal duration. Typically, three orders T=2, 3, and 4 are used sequentially to generate portions of the desired output frequency range using the minimum possible transpose order. If an output with a transpose range higher than 4 is required, it can be generated by frequency shifting. The time domain of the transpose is generated as close as possible to the critical sampling fundamental frequency for processing to minimize computational complexity.

[0033] Harmonic transposers can be based on QMF or DFT. When using a QMF-based harmonic transposer, a modified phase vocoder structure is used in the QMF domain to fully implement bandwidth extension of the core encoder's time-domain signal to perform sampling and subsequent time extension for each QMF subband. Transposition using several transpose factors (e.g., T=2, 3, 4) is implemented in a common QMF analysis / synthesis transform stage. Since QMF-based harmonic transposers do not have the feature of adaptive frequency-domain oversampling, the corresponding flag (sbrOversamplingFlag[ch]) in the bitstream can be ignored.

[0034] When using a DFT-based harmonic transposer, factor 3 and factor 4 transposers (3rd and 4th order transposers) are better integrated into factor 2 transposers (2nd order transposers) by interpolation to reduce complexity. For each frame (corresponding to the coreCoderFrameLength core encoder sample), the nominal "full-form" transform size of the transposer is first determined by the signal adaptive frequency domain oversampling flag (sbrOversamplingFlag[ch]) in the bitstream.

[0035] When sbrPatchingMode == 1 indicates that a linear transpose will be used to generate the high-frequency band, an additional step can be introduced to avoid discontinuous shapes in the spectral envelope of the high-frequency signal input to the subsequent envelope adjuster. This improves the operation of the subsequent envelope adjustment stage, resulting in a perceived more stable high-frequency band signal. The additional preprocessing operation is beneficial for signals with large level variations in the coarse spectral envelope of the low-frequency band signal used for high-frequency reconstruction. However, the value of the bitstream element can be determined in the encoder by applying any kind of signal dependency classification. Preferably, additional preprocessing is initiated via a 1-bit bitstream element bs_sbr_preprocessing. Additional processing is enabled when bs_sbr_preprocessing is set to 1. Additional preprocessing is disabled when bs_sbr_preprocessing is set to 0. The additional processing preferably utilizes a pregain curve used by the high-frequency generator to proportionally adjust each patched low-frequency band XLow. For example, the pregain curve can be calculated according to the following equation: ,0≤k <k0 The first QMF sub-band in the k0 series main frequency band table and the lowEnvSlope series are calculated using a function (such as polyfit()) that computes the coefficients of a best-fit polynomial (in a least-squares sense). For example, a cubic polynomial can be used. ; And among them ,0≤k <k0 Where x_lowband(k) = [0...k0-1], numTimeSlot is the number of SBR envelope time slots existing within a frame, RATE is a constant indicating the number of QMF sub-band samples per time slot (e.g., 2), and φk is a linear prediction filter coefficient (obtainable from the covariance method), and where...

[0036] A bitstream generated according to the MPEG USAC standard (sometimes referred to herein as a "USAC bitstream") contains encoded audio content and typically includes post-processing information indicating various types of spectrum band copying processes applied by a decoder to decode the audio content of the USAC bitstream, and / or post-processing information controlling such spectrum band copying processes and / or indicating at least one feature or parameter of at least one SBR tool and / or eSBR tool used to decode the audio content of the USAC bitstream.

[0037] In this document, we use the term "enhanced SBR post-data" (or "eSBR post-data") to refer to post-data, which indicates various types of spectrum band copying processing applied by a decoder to decode the audio content of an encoded audio bitstream (e.g., a USAC bitstream), and / or controls this spectrum band copying processing, and / or indicates at least one feature or parameter of at least one SBR tool and / or eSBR tool used to decode this audio content but not described or mentioned in the MPEG-4 AAC standard. An example of eSBR post-data is post-data (indicating or used to control spectrum band copying processing) described or mentioned in the MPEG USAC standard but not described or mentioned in the MPEG-4 AAC standard. Therefore, eSBR post-data in this document is not post-data of SBR post-data, and SBR post-data in this document is not post-data of eSBR post-data.

[0038] A USAC bitstream may contain both SBR post-processing data and eSBR post-processing data. More specifically, a USAC bitstream may contain eSBR post-processing data that controls eSBR processing performed by a decoder and SBR post-processing data that controls SBR processing performed by a decoder. According to a typical embodiment of the invention, eSBR post-processing data (e.g., eSBR-specific configuration data) is contained (according to the invention) in an MPEG-4 AAC bitstream (e.g., in the sbr_extension() region at the end of an SBR payload).

[0039] During the decoding of an encoded bitstream using an eSBR toolset (which includes at least one eSBR tool), an eSBR process is performed by a decoder to regenerate the high-frequency band of the audio signal based on a copy of the harmonic sequence truncated during encoding. This eSBR process typically adjusts the spectral envelope of the generated high-frequency band and applies inverse filtering, and adds noise and sinusoidal components to regenerate the spectral characteristics of the original audio signal.

[0040] According to a typical embodiment of the present invention, eSBR post-processing data (e.g., a small number of control bits of eSBR post-processing data) is contained in one or more post-processing data segments of an encoded audio bitstream (e.g., an MPEG-4 AAC bitstream), which also contains encoded audio data in other segments (audio data segments). Typically, at least one of each block of the bitstream is (or contains) a padding element (containing an identifier indicating the start of the padding element), and the eSBR post-processing data is contained in the padding element following the identifier.

[0041] Figure 1 is a block diagram of an exemplary audio processing chain (an audio data processing system), wherein one or more elements of the system can be configured according to an embodiment of the present invention. The system includes the following elements coupled together as shown: encoder 1, transmission subsystem 2, decoder 3, and post-processing unit 4. In variations of the system shown, one or more elements are omitted, or additional audio data processing units are included.

[0042] In some implementations, encoder 1 (which may include a preprocessing unit) is configured to accept PCM (time-domain) samples, including audio content as input, and output an encoded audio bitstream (having a format conforming to the MPEG4 AAC standard) indicating the audio content. The data in the bitstream indicating the audio content is sometimes referred to herein as "audio data" or "encoded audio data." If the encoder is configured according to a typical embodiment of the invention, the audio bitstream output from the encoder includes eSBR post-processing data (and typically also other post-processing data) and audio data.

[0043] It can be verified that one or more encoded audio bitstreams output by autoencoder 1 are transmitted to encoded audio transmission subsystem 2. Subsystem 2 is configured to store and / or transmit each encoded bitstream output by autoencoder 1. An encoded audio bitstream output by autoencoder 1 may be stored by subsystem 2 (e.g., in the form of a DVD or Blu-ray disc), or transmitted by subsystem 2 (which may implement a transmission link or network), or may be both stored and transmitted by subsystem 2.

[0044] Decoder 3 is configured to decode an encoded MPEG-4 AAC audio bitstream (generated by encoder 1) received via subsystem 2. In some embodiments, decoder 3 is configured to extract eSBR post-processing data from each block of the bitstream and decode the bitstream (including performing eSBR processing using the extracted eSBR post-processing data) to produce decoded audio data (e.g., decoding a PCM audio sample stream). In some embodiments, decoder 3 is configured to extract SBR post-processing data from the bitstream (but ignore the eSBR post-processing data contained in the bitstream) and decode the bitstream (including performing SBR processing using the extracted SBR post-processing data) to produce decoded audio data (e.g., decoding a PCM audio sample stream). Typically, decoder 3 includes a buffer that stores (e.g., in a non-transitory manner) segments of the encoded audio bitstream received from subsystem 2.

[0045] The post-processing unit 4 in Figure 1 is configured to receive a decoded audio data stream (e.g., a decoded PCM audio sample) from one of the decoders 3 and perform post-processing on it. The post-processing unit can also be configured to reproduce the post-processed audio content (or the decoded audio received from the decoder 3) for playback by one or more speakers.

[0046] Figure 2 is a block diagram of an encoder (100), which is an embodiment of the audio processing unit of the present invention. Any component or element of the encoder 100 may be implemented as one or more programs and / or one or more circuits (e.g., ASIC, FPGA, or other integrated circuits) in hardware, software, or a combination of both. The encoder 100 includes an encoder 105, a filler / formatter stage 107, a post-data generation stage 106, and a buffer memory 109 connected as shown. Typically, the encoder 100 also includes other processing elements (not shown). The encoder 100 is configured to convert an input audio bitstream into an encoded output MPEG-4 AAC bitstream.

[0047] The post-data generator 106 is coupled and configured to generate post-data (including eSBR post-data and SBR post-data) (and / or transfer post-data to stage 107) so that it is included in the encoded bit stream output by the autoencoder 100 by stage 107.

[0048] Encoder 105 is coupled and configured to encode input audio data (e.g., by performing compression on it) and verifies the resulting encoded audio to stage 107 to be included in the encoded bit stream output from stage 107.

[0049] Stage 107 is configured to multiplex encoded audio from encoder 105 and post-processing data (including eSBR post-processing data and SBR post-processing data) from generator 106 to generate an encoded bitstream output from stage 107, preferably such that the encoded bitstream has a format specified by one embodiment of the present invention.

[0050] The buffer memory 109 is configured to store (e.g., in a non-transitory manner) at least one block of the encoded audio bitstream output from stage 107, and then confirms a sequence of blocks of the encoded audio bitstream as output from the encoder 100 from the buffer memory 109 to a transmission system.

[0051] Figure 3 is a block diagram of a system that includes a decoder (200) (which is one embodiment of the audio processing unit of the present invention) and, where appropriate, a post-processor (300) coupled to the decoder 200. Any component or element of the decoder 200 and the post-processor 300 may be implemented in hardware, software, or a combination of both as one or more programs and / or one or more circuits (e.g., ASIC, FPGA, or other integrated circuits). The decoder 200 includes a buffer memory 201, a bitstream payload deformatter (parser) 205, an audio decoding subsystem 202 (sometimes referred to as a "core" decoding stage or "core" decoding subsystem), an eSBR processing stage 203, and a control bit generation stage 204, all connected as shown. Typically, the decoder 200 also includes other processing elements (not shown).

[0052] Buffer memory (buffer) 201 stores (e.g., in a non-transitory manner) at least one block of an encoded MPEG-4 AAC audio bitstream received by decoder 200. During the operation of decoder 200, a sequence of blocks of the bitstream is verified from buffer 201 to deformatter 205.

[0053] In a variation of the embodiment of Figure 3 (or the embodiment of Figure 4 to be described), an APU (which is not a decoder) (e.g., APU 500 of Figure 6) includes a buffer memory (e.g., a buffer memory identical to buffer 201) that stores (e.g., in a non-transitory manner) at least one block of an encoded audio bitstream of the same type (e.g., an MPEG-4 AAC audio bitstream) received by buffer 201 of Figure 3 or Figure 4 (i.e., an encoded audio bitstream containing eSBR post-processing data).

[0054] Referring again to Figure 3, the deformatter 205 is coupled and configured to decompose the blocks of the multi-bit stream to extract SBR post-processing data (including quantization envelope data) and eSBR post-processing data (and usually other post-processing data) to verify that at least the eSBR post-processing data and SBR post-processing data are delivered to the eSBR processing stage 203, and usually also to the decoding subsystem 202 (and, depending on the case, to the control bit generator 204). The deformatter 205 is also coupled and configured to extract audio data from the blocks of the bit stream and verify that the extracted audio data is delivered to the decoding subsystem (decoding stage) 202.

[0055] The system in Figure 3 may also include a post-processor 300. The post-processor 300 includes a buffer memory (buffer) 301 and other processing elements (not shown), including at least one processing element coupled to the buffer 301. The buffer 301 stores (e.g., in a non-transitory manner) at least one block (or frame) of decoded audio data received by the post-processor 300 from the decoder 200. The processing elements of the post-processor 300 are coupled and configured to receive and use the post-processor data output by the self-decoding subsystem 202 (and / or deformatter 205) and / or the control bits output by the stage 204 of the self-decoding 200 to adapt the processing of a sequence of blocks (or frames) of decoded audio output from the buffer 301.

[0056] The audio decoding subsystem 202 of decoder 200 is configured to decode the audio data extracted by parser 205 (this decoding can be referred to as a "core" decoding operation) to generate decoded audio data and verify the decoded audio data to eSBR processing stage 203. Decoding is performed in the frequency domain and typically includes inverse quantization followed by spectral processing. Typically, one of the final processing stages in subsystem 202 applies a frequency-to-time domain transform to the decoded frequency-domain audio data, such that the subsystem output is time-domain decoded audio data. Stage 203 is configured to apply the SBR tools and eSBR tools indicated by eSBR post-processing data and eSBR (extracted by parser 205) to the decoded audio data (i.e., using SBR and eSBR post-processing data to perform SBR and eSBR processing on the output of decoding subsystem 202) to generate fully decoded audio data from the output of decoder 200 (e.g., to post-processor 300). Typically, decoder 200 includes a memory (accessible by subsystem 202 and stage 203) storing deformatted audio data and post-processing data from the output of deformatter 205. Stage 203 is configured to access audio data and post-processing data (including SBR and eSBR post-processing data) as needed during SBR and eSBR processing. The SBR and eSBR processing in stage 203 can be considered as post-processing of the output of the core decoding subsystem 202. Decoder 200 may also include, where appropriate, a final upmixing subsystem (which can use PS post-processing data extracted from deformatter 205 and / or control bits generated in subsystem 204 to apply the parametric stereo ("PS") tool defined in the MPEG-4 AAC standard), which is coupled and configured to upmix the output of stage 203 to produce a fully decoded upmixed audio output from decoder 200. Alternatively, the post-processor 300 is configured to perform upmixing on the output of the decoder 200 (e.g., using PS post-processor data extracted by the deformatter 205 and / or control bits generated in the subsystem 204).

[0057] In response to post-processing data extracted by deformatter 205, control bit generator 204 can generate control data, which can be used within decoder 200 (e.g., in a final upmixing subsystem) and / or validated as output of decoder 200 (e.g., to postprocessor 300 for post-processing). In response to post-processing data extracted from the input bitstream (and, where appropriate, control data), stage 204 can generate control bits (and validate control bits to postprocessor 300) to indicate that decoded audio data output from eSBR processing stage 203 should undergo a specific type of post-processing. In some embodiments, decoder 200 is configured to validate post-processing data extracted from the input bitstream by deformatter 205 to postprocessor 300, and postprocessor 300 is configured to use the post-processing data to perform post-processing on decoded audio data output from decoder 200.

[0058] Figure 4 is a block diagram of an audio processing unit (“APU”) (210), which is another embodiment of the audio processing unit of the present invention. APU 210 is a legacy decoder that performs eSBR processing unconfigured. Any component or element of APU 210 may be implemented in hardware, software, or a combination of hardware and software as one or more programs and / or one or more circuits (e.g., ASIC, FPGA, or other integrated circuits). APU 210 includes a buffer memory 201, a bitstream payload deformatter (parser) 215, an audio decoding subsystem 202 (sometimes referred to as a “core” decoding stage or “core” decoding subsystem), and an SBR processing stage 213, connected as shown. Typically, APU 210 also includes other processing elements (not shown). APU 210 may represent, for example, an audio encoder, decoder, or transcoder.

[0059] Elements 201 and 202 of APU 210 are identical to the numbered elements of decoder 200 (Figure 3), and their descriptions above will not be repeated. In operation of APU 210, a block sequence of an encoded audio bitstream (an MPEG-4 AAC bitstream) received by APU 210 is verified from buffer 201 to deformatter 215.

[0060] Deformatter 215 is coupled and configured to decompose each block of the multi-bit stream to extract SBR post-processing data (including quantization envelope data) from it, and typically also extracts other post-processing data from it, but ignores eSBR post-processing data that may be included in the bit stream according to any embodiment of the present invention. Deformatter 215 is configured to verify at least SBR post-processing data to SBR processing stage 213. Deformatter 215 is also coupled and configured to extract audio data from each block of the bit stream and verify the extraction of audio data to decoding subsystem (decoding stage) 202.

[0061] The audio decoding subsystem 202 of decoder 200 is configured to decode the audio data extracted by deformatter 215 (this decoding may refer to a "core" decoding operation) to generate decoded audio data and verify the decoded audio data to SBR processing stage 213. Decoding is performed in the frequency domain. Typically, one of the final processing stages in subsystem 202 applies a frequency-to-time domain transformation to the decoded frequency-domain audio data, such that the subsystem output is time-domain decoded audio data. Stage 213 is configured to apply an SBR tool (but not an eSBR tool) indicated by SBR post-processor data (extracted by deformatter 215) to the decoded audio data (i.e., using SBR post-processor data to perform SBR processing on the output of decoding subsystem 202) to generate fully decoded audio data from APU 210 output (e.g., to post-processor 300). Typically, the APU 210 includes a memory (accessible by subsystem 202 and stage 213) storing deformatted audio data and post-processing data from the output of deformatter 215. Stage 213 is configured to access audio data and post-processing data (including SBR post-processing data) as needed during SBR processing. The SBR processing in stage 213 can be considered as post-processing of the output of the core decoding subsystem 202. The APU 210 may also include, as appropriate, a final upmixing subsystem (which can use the PS post-processing data extracted by deformatter 215 to apply the parametric stereo "PS" tool defined in the MPEG-4 AAC standard), which is coupled and configured to perform upmixing on the output of stage 213 to produce a fully decoded upmixed audio output from the APU 210. Alternatively, a post-processor is configured to perform upmixing on the output of APU 210 (e.g., using PS post-processor data extracted by deformatter 215 and / or control bits generated in APU 210).

[0062] Various implementations of encoder 100, decoder 200 and APU 210 are configured to perform different embodiments of the method of the present invention.

[0063] According to some embodiments, eSBR post-processing data (e.g., a small number of control bits of eSBR post-processing data) is included in an encoded audio bitstream (e.g., an MPEG-4 AAC bitstream), allowing legacy decoders (which parse the eSBR post-processing data unconfigured or use any eSBR tools associated with it) to ignore the eSBR post-processing data but still decode the bitstream as much as possible without using the eSBR post-processing data or any eSBR tools associated with it, typically without significant loss of decoded audio quality. However, eSBR decoders (which are configured to parse the bitstream to identify the eSBR post-processing data and respond to it using at least one eSBR tool) will benefit from using at least one of these eSBR tools. Therefore, embodiments of the present invention provide a method for efficiently transmitting enhanced spectrum band replication (eSBR) control data or post-processing data in a backtracking compatible manner.

[0064] Typically, the eSBR post-data in the bitstream indicates one or more of the following eSBR tools (e.g., indicating at least one feature or parameter of one or more of the following eSBR tools) (these eSBR tools are described in the MPEG USAC standard and may or may not be applied by an encoder during the generation of the bitstream): Harmonic transpose; and QMF patching additional preprocessing (pre-flattening).

[0065] For example, the eSBR post-processing data contained in the bitstream can indicate the values ​​of parameters (as described in the MPEG USAC standard and in this invention): sbrPatchingMode[ch], sbrOversamplingFlag[ch], sbrPitchInBins[ch], sbrPitchInBins[ch], and bs_sbr_preprocessing.

[0066] In this paper, the symbol X[ch] (where X is a parameter) indicates that the parameter is related to a channel ("ch") of the audio content of an encoded bitstream to be decoded. For simplicity, we sometimes omit the expression [ch] and assume that the relevant parameter is related to a channel of the audio content.

[0067] In this paper, the symbol X[ch][env] (where X is a parameter) indicates that the parameter is related to the SBR envelope ("env") of the channel ("ch") of the audio content of the encoded bitstream to be decoded. For simplicity, we sometimes omit the expressions [env] and [ch], and assume that the relevant parameter is related to the SBR envelope of one channel of the audio content.

[0068] During the decoding of an encoded bitstream, harmonic transposition (for each channel "ch" of the audio content indicated by the bitstream) is performed during one of the eSBR processing stages of the decoding process. This is controlled by the following eSBR post-data parameters: sbrPatchingMode[ch], sbrOversamplingFlag[ch], sbrPitchInBinsFlag[ch], and sbrPitchInBins[ch].

[0069] The value "sbrPatchingMode[ch]" indicates the transpose type used in the eSBR: sbrPatchingMode[ch]=1 indicates linear transpose patching as described in section 4.6.18 of the MPEG-4 AAC standard (used with high-quality SBR or low-power SBR); sbrPatchingMode[ch]=0 indicates harmonic SBR patching as described in section 7.5.3 or 7.5.4 of the MPEG USAC standard.

[0070] The value "sbrOversamplingFlag[ch]" indicates that adaptive frequency domain oversampling in eSBR is used in combination with DFT-based harmonic SBR patching as described in section 7.5.3 of the MPEG USAC standard. This flag controls the size of the DFT in the transposer: 1 indicates that adaptive frequency domain oversampling is enabled as described in section 7.5.3.1 of the MPEG USAC standard; 0 indicates that adaptive frequency domain oversampling is disabled as described in section 7.5.3.1 of the MPEG USAC standard.

[0071] The value "sbrPitchInBinsFlag[ch]" controls the interpretation of the sbrPitchInBins[ch] parameter: 1 indicates that the value of sbrPitchInBins[ch] is valid and greater than 0; 0 indicates that the value of sbrPitchInBins[ch] is set to 0.

[0072] The value "sbrPitchInBins[ch]" controls the addition of the cross-product terms in the SBR harmonic transposer. The value sbrPitchinBins[ch] is an integer value in the range [0, 127] and represents the distance measured in a frequency grid of a 1536-line DFT at the sampling frequency of the core encoder.

[0073] If an MPEG-4 AAC bitstream indicates that its channels are not coupled to one SBR channel pair (rather than a single SBR channel), then the bitstream indicates two instances of the above syntax (for harmonic or non-harmonic transpose): one instance of each channel, sbr_channel_pair_element().

[0074] Harmonic transposition in eSBR tools typically improves the quality of decoded music signals at relatively low crossover frequencies. Non-harmonic transposition (i.e., older spectral patching) typically enhances speech signals. Therefore, one of the bases for determining which type of transposition is more suitable for encoding specific audio content is to select the transposition method based on speech / music detection with harmonic transposition applied to the music content and spectral patching for the tempo content.

[0075] Pre-flattening during eSBR processing is controlled by the value of a unit eSBR post-data parameter called "bs_sbr_preprocessing," which, in a sense, determines whether pre-flattening is performed or not. When using the SBR QMF patching algorithm described in section 4.6.18.6.3 of the MPEG-4 AAC standard, a pre-flattening step (when indicated by the "bs_sbr_preprocessing" parameter) can be performed to attempt to avoid discontinuities in the shape of the spectral envelope of a high-frequency signal input to a subsequent envelope adjuster (another stage of eSBR processing). Pre-flattening typically improves the operation of the subsequent envelope adjustment stage, resulting in a perceived more stable high-frequency band signal.

[0076] According to some embodiments of the present invention, the total bit rate requirement included in the MPEG-4 AAC bitstream eSBR post-processing data of one of the aforementioned eSBR tools (harmonic transpose and pre-flattening) is expected to be several hundred bits per second, since only differential control data required to perform eSBR processing is transmitted. Legacy decoders can ignore this information because it is included in a backtracking-compatible manner (as explained later). Therefore, the adverse effect on the bit rate associated with including the eSBR post-processing data is negligible for several reasons: The bit rate loss (attributed to the inclusion of eSBR post-processing data) accounts for a very small percentage of the total bit rate because only the differential control data required for performing eSBR processing (and is not part of the SBR control data cascade) is transmitted; and The tuning of SBR-related control information typically does not depend on the transpose details. Examples of control data depending on the operation of the transposer will be discussed later in this application.

[0077] Therefore, embodiments of the present invention provide a method for efficiently transmitting enhanced spectrum band replication (eSBR) control data or meta data in a backtracking-compatible manner. This efficient transmission of eSBR control data reduces memory requirements in decoders, encoders, and transcoders employing the present invention, without materially adversely affecting the bit rate. Furthermore, it reduces the complexity and processing requirements associated with implementing eSBR according to embodiments of the present invention, since SBR data only needs to be processed once and is not rebroadcast, as is the case when eSBR is treated as a completely independent object type in MPEG-4 AAC rather than being integrated into the MPEG-4 AAC codec in a backtracking-compatible manner.

[0078] Next, referring to FIG7, we describe the elements of a block ("raw_data_block") of an MPEG-4 AAC bitstream (containing eSBR post-processing data) according to some embodiments of the present invention. FIG7 is a diagram of a block ("raw_data_block") of an MPEG-4 AAC bitstream, which shows some segments of the MPEG-4 AAC bitstream.

[0079] A block of an MPEG-4 AAC bitstream may contain at least one "single_channel_element()" (e.g., a single-channel element shown in Figure 7) and / or at least one "channel_pair_element()" (not explicitly shown in Figure 7, but it may exist), which contains audio data of an audio program. The block may also contain several "fill_element" (e.g., fill element 1 and / or fill element 2 in Figure 7), which contain program-related data (e.g., post-processing data). Each "single_channel_element()" contains an identifier indicating the start of a single-channel element (e.g., "ID1" in Figure 7) and may contain audio data indicating different channels of a multi-channel audio program. Each "channel_pair_element" contains an identifier indicating the start of a channel pair element (not shown in Figure 7) and may contain audio data indicating the two channels of the program.

[0080] A fill element (referred to herein as a fill element) in an MPEG-4 AAC bitstream contains an identifier (“ID2” in Figure 7) indicating the start of a fill element and the fill data following the identifier. The identifier ID2 may consist of a 3-bit unsigned integer with a value of 0×6, transmitting the most significant bit first (“uimsbf”). The fill data may contain an extension_payload() element (sometimes referred to herein as an extended payload) whose syntax is shown in Table 4.57 of the MPEG-4 AAC standard. Several types of extended payloads exist and are identified by the “extension_type” parameter, which is a 4-bit unsigned integer, transmitting the most significant bit first (“uimsbf”).

[0081] The padding data (such as one of its extended payloads) may contain a header or identifier (e.g., "Header 1" in Figure 7) that indicates a segment of the padding data (which indicates an SBR object) (i.e., the header initializes one of the "SBR object" types referred to in the MPEG-4 AAC standard sbr_extension_data()). For example, the value of "1101" or "1110" in the extension_type field of the header is used to identify a spectrum band copy (SBR) extended payload, where the identifier "1101" identifies an extended payload with SBR data and "1110" identifies an extended payload containing SBR data with a cyclic redundancy check (CRC) to verify the correctness of the SBR data.

[0082] When a header (e.g., the extension_type field) initializes an SBR object type, the SBR post-data (sometimes referred to herein as "bandwidth copy data," and specifically as sbr_data() in the MPEG-4 AAC standard) follows the header, and at least one bandwidth copy extension element (e.g., the "SBR extension element" in padding element 1 of Figure 7) may follow the SBR post-data. This bandwidth copy extension element (a segment of the bitstream) refers to an "sbr_extension()" region in the MPEG-4 AAC standard. A bandwidth copy extension element may include a header (e.g., the "SBR extension header" in padding element 1 of Figure 7).

[0083] The MPEG-4 AAC standard anticipates that a spectrum copy extension element can contain PS (parametric stereo) data for audio data in a program. The MPEG-4 AAC standard anticipates that when the header of a padding element (e.g., one of its extended payloads) initializes an SBR object type (as shown in "Header 1" in Figure 7) and one of the padding elements' spectrum copy extension elements contains PS data, the padding element (e.g., its extended payload) contains spectrum copy data and a "bs_extension_id" parameter (whose value (i.e., bs_extension_id=2) indicates that the PS data is contained within one of the padding elements' spectrum copy extension elements).

[0084] According to some embodiments of the present invention, eSBR post-processing data (e.g., a flag indicating whether enhanced spectrum band replication (eSBR) processing is performed on the audio content of a block) is included in a spectrum band replication extension element of a padding element. For example, this flag is indicated in padding element 1 of FIG. 7, where the flag appears after the header of the "SBR extension element" of padding element 1 (the "SBR extension header" of padding element 1). This flag and additional eSBR post-processing data are included, as appropriate, in the spectrum band replication extension element after the header of the spectrum band replication extension element (e.g., in the SBR extension element of padding element 1 in FIG. 7 after the SBR extension header). According to some embodiments of the present invention, a padding element containing eSBR post-processing data also includes a "bs_extension_id" parameter, the value of which (e.g., bs_extension_id=3) indicates that eSBR post-processing data is included in the padding element and that eSBR processing is performed on the audio content of the relevant block.

[0085] According to some embodiments of the present invention, the eSBR post-processing data is contained in a padding element of an MPEG-4 AAC bitstream (e.g., padding element 2 in FIG. 7) rather than in a spectral band copy extension element (SBR extension element). This is because a padding element containing an extension_payload() (which has SBR data or SBR data with a CRC) does not contain any other extension payload of any other extension type. Therefore, in embodiments where the eSBR post-processing data stores its own extension payload, a separate padding element is used to store the eSBR post-processing data. This padding element contains an identifier (e.g., "ID2" in FIG. 7) indicating the start of a padding element and padding data following the identifier. The padding data may contain an extension_payload() element (sometimes referred to herein as an extension payload) whose syntax is shown in Table 4.57 of the MPEG-4 AAC standard. The padding data (e.g., one of its extended payloads) includes a header indicating an eSBR object (e.g., "Header 2" of padding element 2 in Figure 7) (i.e., the header initializes an Enhanced Spectral Band Replication (eSBR) object type), and the padding data (e.g., one of its extended payloads) includes eSBR post-processing data following the header. For example, padding element 2 in Figure 7 includes this header ("Header 2") and also includes eSBR post-processing data following the header (i.e., the "flag" in padding element 2, which indicates whether Enhanced Spectral Band Replication (eSBR) processing is performed on the audio content of the block). Additional eSBR post-processing data may also be included, as appropriate, in the padding data of padding element 2 in Figure 7 following header 2. In the embodiments described in this paragraph, the header (e.g., header 2 in Figure 7) has an identification value that is not a conventional value specified in Table 4.57 of the MPEG-4 AAC standard, but instead indicates an eSBR extension payload (such that the extension_type field of the header indicates that the fill data includes eSBR post-data).

[0086] In a first-type embodiment, the present invention is an audio processing unit (e.g., a decoder) comprising: A memory (e.g., buffer 201 in Figure 3 or 4) is configured to store at least one block of an encoded audio bitstream (e.g., at least one block of an MPEG-4 AAC bitstream); A bitstream payload deformatter (e.g., element 205 of Figure 3 or element 215 of Figure 4), coupled to the memory and configured to demultiplex at least a portion of the block of the bitstream; and A decoding subsystem (e.g., elements 202 and 203 of Figure 3 or elements 202 and 213 of Figure 4) is coupled and configured to decode at least one portion of the audio content of a block of the bitstream, wherein the block comprises: A padding element comprising an identifier indicating the start of the padding element (e.g., an "id_syn_ele" identifier with the value 0x6 in Table 4.85 of the MPEG-4 AAC standard) and padding data following the identifier, wherein the padding data comprises: At least one flag identifies whether enhanced spectrum band replication (eSBR) processing is performed on the audio content of the block (e.g., using spectrum band replication data and eSBR post-data contained in the block).

[0087] This flag is eSBR post-processing data, and one instance of this flag is the sbrPatchingMode flag. Another instance of this flag is the harmonicSBR flag. Both of these flags indicate either a basic form of spectrum band copying or an enhanced form of spectrum copying for the audio data in this block. The basic form of spectrum copying is spectrum patching, and the enhanced form of spectrum band copying is harmonic transpose.

[0088] In some embodiments, the padding data also includes additional eSBR post-data (i.e., eSBR post-data other than the flag).

[0089] The memory may be a buffer memory (e.g., one embodiment of buffer 201 in FIG4) that stores (e.g., in a non-transitory manner) at least one block of the encoded audio bitstream.

[0090] It is estimated that the complexity of performing eSBR processing (using eSBR harmonic transpose and pre-flattening) by an eSBR decoder during the decoding of one of the MPEG-4 AAC bitstreams containing eSBR post-processing data (indicating such eSBR tools) will be as follows (for typical decoding with indicator parameters): Harmonic transpose (16 kbps, 14400 / 28800 Hz) ○ Based on DFT: 3.68 WMOPS (weighted millions of operations per second); ○ Based on QMF: 0.98 WMOPS; QMF patch pretreatment (pre-flattening): 0.1 WMOPS. As is well known, for transient states, DFT-based transpose generally performs better than QMF-based transpose.

[0091] According to some embodiments of the present invention, a padding element (of an encoded audio bitstream) containing eSBR post-processing data also includes a parameter (e.g., a "bs_extension_id" parameter) whose value (e.g., bs_extension_id=3) indicates that eSBR post-processing data is contained in the padding element and that eSBR processing is performed on the audio content of the relevant block, and / or a parameter (e.g., the same "bs_extension_id" parameter) whose value (e.g., bs_extension_id=2) indicates that one sbr_extension() region of the padding element contains PS data. For example, as indicated in Table 1 below, this parameter having a value of bs_extension_id=2 indicates that one sbr_extension() region of the padding element contains PS data, and this parameter having a value of bs_extension_id=3 indicates that one sbr_extension() region of the padding element contains eSBR post-processing data. Table 1 [bs_extension_id] [meaning] [Han] 0 reserve 1 reserve 2 EXTENSION_ID_PS 3 EXTENSION_ID_ESBR

[0092] According to some embodiments of the present invention, the syntax of each band replication extension element containing eSBR post-data and / or PS data is indicated in Table 2 below (where "sbr_extension()" represents a region of the band replication extension element, "bs_extension_id" is as described in Table 1 above, "ps_data" represents PS data, and "esbr_data" represents eSBR post-data): Table 2 sbr_extension(bs_extension_id, num_bits_left) { switch (bs_extension_id) { case EXTENSION_ID_PS: num_bits_left -= ps_data(); Note 1 break case EXTENSION_ID_ESBR: num_bits_left -= esbr_data(); Note 2 break default: [ bs_fill_bits]; [ ] num_bits_left = 0; break } } Note 1: ps_data() returns the number of bits read. Note 2: esbr_data() returns the number of bits read. In one exemplary embodiment, esbr_data() mentioned in Table 2 above indicates the value of the following meta data parameter: 1. The unit cell is followed by the data parameter "bs_sbr_preprocessing"; and 2. For each channel ("ch") of the audio content of the encoded bitstream to be decoded, the above parameters are "sbrPatchingMode[ch]", "SbrOversamplingFlag[ch]", "SbrPitchInBinsFlag[ch]" and "sbrPitchInBins[ch]".

[0093] For example, in some embodiments, esbr_data() may have the syntax indicated in Table 3 to indicate such meta data parameters: Table 3 grammar Number of bits esbr_data(id_aac, bs_coupling) { [bs_sbr_preprocessing]; [1] [ ]if (id_aac == ID_SCE) { if ( [sbrPatchingMode[0] ]== 0) { [1] [sbrOversamplingFlag[0]] [;] [1] if ( [sbrPitchInBinsFlag[0]]) [1] [ sbrPitchInBins[0]]; [7] else sbrPitchInBins[0] = 0; } else { sbrOversamplingFlag[0] = 0; sbrPitchInBins[0] = 0; } } else if (id_aac == ID_CPE) { If (bs_coupling) { if ( [sbrPatchingMode[0,1] ]== 0) { [1] [ ] [sbrOversamplingFlag[0,1]] [;] [1] if ( [sbrPitchInBinsFlag[0,1]]) [1] [ ] [ sbrPitchInBins[0,1]]; [7] else sbrPitchInBins[0,1] = 0; } else { sbrOversamplingFlag[0,1] = 0; sbrPitchInBins[0,1] = 0; } } else { / * bs_coupling == 0 * / if ( [sbrPatchingMode[0] ]== 0) { [1] [ ] [ sbrOversamplingFlag[0]] [;] [1] if ( [sbrPitchInBinsFlag[0]]) [1] [ ] [ sbrPitchInBins[0]]; [7] else sbrPitchInBins[0] = 0; } else { sbrOversamplingFlag[0] = 0; sbrPitchInBins[0] = 0; } if ( [sbrPatchingMode[1] ]== 0) { [1] [ ] [ sbrOversamplingFlag[1]] [;] [1] if ( [sbrPitchInBinsFlag[1]]) [1] [ ] [ sbrPitchInBins[1]]; [7] else sbrPitchInBins[1] = 0; } else { sbrOversamplingFlag[1] = 0; sbrPitchInBins[1] = 0; } } } } Note: bs_sbr_preprocessing is defined as described in section 6.2.12 of ISO / IEC 23003-3:2012. sbrPatchingMode[ch], sbrOversamplingFlag[ch], sbrPitchInBinsFlag[ch], and sbrPitchInBins[ch] are defined as described in section 7.5 of ISO / IEC 23003-3:2012.

[0094] The above syntax enables the efficient implementation of one enhancement form of spectral band copying (such as harmonic transpose) as an extension of a legacy decoder. Specifically, the eSBR data in Table 3 only contains the parameters required to perform the enhancement form of spectral band copying, which are no longer supported in the bitstream and cannot be directly derived from the parameters already supported in the bitstream. All other parameters required to perform the enhancement form of spectral band copying and the processing data are extracted from readily available parameters in defined locations within the bitstream.

[0095] For example, an MPEG-4 HE-AAC or HE-AAC v2 compatible decoder can be extended to include an enhanced form of spectral band copying, such as harmonic transpose. This enhanced form of spectral band copying is an addition to the basic form of spectral band copying already supported by the decoder. In the context of an MPEG-4 HE-AAC or HE-AAC v2 compatible decoder, this basic form of spectral band copying is the QMF Spectral Patching SBR tool, as defined in section 4.6.18 of the MPEG-4 AAC standard.

[0096] When performing the enhanced form of spectrum band replication, an extended HE-AAC decoder can reuse many bitstream parameters already included in the SBR extended payload of the bitstream. Specific parameters that can be reused include, for example, various parameters for determining the master band table. These parameters include bs_start_freq (the parameter for determining the start of the master band table parameters), bs_stop_freq (the parameter for determining the stop of the master band table), bs_freq_scale (the parameter for determining the number of bands per octave), and bs_alter_scale (the parameter for the scale of band alteration). Reusable parameters also include parameters for determining the noise band table (bs_noise_bands) and the limiter band table (bs_limiter_bands). Therefore, in various embodiments, at least some equivalent parameters specified in the USAC standard are omitted from the bitstream to reduce the control burden on the bitstream. Typically, when a parameter specified in the AAC standard has an equivalent parameter specified in the USAC standard, the equivalent parameter specified in the USAC standard has the same name as the parameter specified in the AAC standard, such as the envelope scaling factor EOrigMapped. However, the equivalent parameter specified in the USAC standard usually has a different value, which is "tuned" according to the enhanced SBR processing defined in the USAC standard rather than the SBR processing defined in the AAC standard.

[0097] It is recommended to enable enhanced SBR to improve the subjective quality of audio content with harmonic frequency structures and strong tonal characteristics, especially at low bit rates. The value of the corresponding bitstream element (i.e., esbr_data()) controlling these tools can be determined by applying a signal dependency classification mechanism within the encoder. Generally, the use of harmonic patching (sbrPatchingMode==1) is more suitable for encoding music signals at very low bit rates, where the audio bandwidth of the core codec is significantly limited. This is particularly prominent when such signals contain a pronounced harmonic structure. Conversely, the use of conventional SBR patching is more suitable for speech and mixed signals because it provides better preservation of one of the temporal structures of speech.

[0098] To improve the performance of the harmonic transposer, a preprocessing step (bs_sbr_preprocessing==1) can be initiated, which attempts to avoid introducing spectral discontinuities into the signal entering the subsequent envelope adjuster. The operation of this tool is beneficial for revealing signal types with large level variations in the coarse spectral envelope of the low-frequency band signal used for high-frequency reconstruction.

[0099] To improve the transient response of harmonic SBR patching, adaptive frequency domain oversampling (sbrOversamplingFlag==1) can be applied. Since adaptive frequency domain oversampling increases the computational complexity of the transposer, but only benefits frames containing transients, the use of this tool is controlled by bit stream elements, which are transmitted once per frame and per independent SBR channel.

[0100] In the proposed enhanced SBR mode, one of the decoders typically needs to be able to switch between legacy SBR patch and enhanced SBR patch. Therefore, depending on the decoder settings, a delay as long as the duration of a core audio frame can be introduced. Generally, the delays for legacy SBR patch and enhanced SBR patch will be similar.

[0101] In addition to numerous parameters, other data elements can also be reused by an extended HE-AAC decoder when performing an enhanced form of spectrum band replication according to an embodiment of the present invention. For example, envelope data and noise limit data can also be extracted from bs_data_env (envelope scaling factor) and bs_noise_env (noise limit scaling factor) data and used during the enhanced form of spectrum band replication.

[0102] Essentially, these embodiments utilize configuration parameters and envelope data already supported by an older HE-AAC or HE-AAC v2 decoder in the SBR extended payload to enable an enhanced form of spectral band replication, requiring as little additional data transmission as possible. The post-processing data is initially tuned according to a basic form of HFR (e.g., spectral shift operation of SBR), but according to embodiments, it is used for an enhanced form of HFR (e.g., harmonic transpose of eSBR). As previously discussed, post-processing data generally refers to operating parameters (e.g., envelope scaling factor, noise floor scaling factor, time / frequency grid parameters, sine wave addition information, variable crossover frequency / band, inverse filtering mode, envelope resolution, smoothing mode, frequency interpolation mode) tuned and designed for use with a basic form of HFR (e.g., linear spectral shift). However, this post-processing data can then be combined with additional post-processing data parameters specifically for an enhanced form of HFR (e.g., harmonic transpose) to efficiently and effectively process audio data using an enhanced form of HFR.

[0103] Therefore, an extended decoder supporting one form of spectrum band replication can be generated in a very efficient manner by relying on predefined bitstream elements (e.g., bitstream elements in an SBR extended payload) and adding only the parameters required to support the enhanced form of spectrum band replication (in a padding extended payload). This data reduction feature, combined with the placement of the newly added parameters in a reserved data field (such as an extended capacity), substantially reduces the barriers to generating a decoder that supports the enhanced form of spectrum band replication by ensuring backward compatibility of the bitstream with legacy decoders that do not support it.

[0104] In Table 3, the numbers in the right row indicate the number of bits of the corresponding parameter in the left row.

[0105] In some embodiments, the SBR object type defined in MPEG-4 AAC is updated to include the SBR tool and the enhanced SBR (eSBR) tool, as indicated in the SBR extension element (bs_extension_id==EXTENSION_ID_ESBR). If a decoder detects and supports this SBR extension element, the decoder adopts the indicated state of the enhanced SBR tool. The SBR object type updated in this manner is referred to as SBR enhancement.

[0106] In some embodiments, the present invention is a method comprising the step of encoding audio data to generate a coded bitstream (e.g., an MPEG-4 AAC bitstream), comprising including eSBR post-processing data in at least one segment of at least one block of the coded bitstream and audio data in at least another segment of the block. In a typical embodiment, the method comprises the step of multiplexing the audio data and the eSBR post-processing data in each block of the coded bitstream. In a typical decoding of the coded bitstream in an eSBR decoder, the decoder extracts the eSBR post-processing data from the bitstream (including parsing and demultiplexing the eSBR post-processing data and the audio data) and uses the eSBR post-processing data to process the audio data to generate a decoded audio data stream.

[0107] Another aspect of the present invention is an eSBR decoder configured to perform eSBR processing (e.g., using at least one of eSBR tools called harmonic transpose or pre-flattening) during the decoding of an encoded audio bitstream (e.g., an MPEG-4 AAC bitstream) that does not contain eSBR post-processing data. An example of such a decoder will be described with reference to FIG5.

[0108] The eSBR decoder (400) of Figure 5 includes a buffer memory 201 (identical to the memory 201 in Figures 3 and 4), a bitstream payload deformatter 215 (identical to the deformatter 215 in Figure 4), an audio decoding subsystem 202 (sometimes referred to as a "core" decoding stage or "core" decoding subsystem, and identical to the core decoding subsystem 202 in Figure 3), an eSBR control data generation subsystem 401, and an eSBR processing stage 203 (identical to stage 203 in Figure 3). Typically, the decoder 400 also includes other processing elements (not shown).

[0109] In the operation of decoder 400, a block sequence of an encoded audio bitstream (an MPEG-4 AAC bitstream) received by decoder 400 is confirmed from buffer 201 to deformatter 215.

[0110] Deformatter 215 is coupled and configured to decompose the blocks of the multi-bit stream to extract SBR post-processing data (including quantization envelope data) and typically other post-processing data. Deformatter 215 is configured to verify at least SBR post-processing data to the eSBR processing stage 203. Deformatter 215 is also coupled and configured to extract audio data from the blocks of the bit stream and verify the extracted audio data to the decoding subsystem (decoding stage) 202.

[0111] The audio decoding subsystem 202 of decoder 400 is configured to decode the audio data extracted by deformatter 215 (this decoding can be referred to as a "core" decoding operation) to generate decoded audio data and verify the decoded audio data to eSBR processing stage 203. Decoding is performed in the frequency domain. Typically, one of the final processing stages in subsystem 202 applies a frequency-to-time domain transformation to the decoded frequency-domain audio data, such that the output of the subsystem is time-domain decoded audio data. Stage 203 is configured to apply the SBR tools (and eSBR tools) indicated by the SBR post-processing data (extracted by deformatter 215) and the eSBR post-processing data generated in subsystem 401 to the decoded audio data (i.e., using the SBR and eSBR post-processing data to perform SBR and ESBR processing on the output of decoding subsystem 202) to generate fully decoded audio data from the output of decoder 400. Typically, decoder 400 includes a memory (accessible by subsystem 202 and stage 203) storing deformatted audio data and post-processing data from deformatter 215 (and, depending, subsystem 401), and stage 203 is configured to access audio data and post-processing data as needed during SBR and eSBR processing. SBR processing in stage 203 can be considered as post-processing of the output of core decoding subsystem 202. Decoder 400 also includes, depending on the case, a final upmixing subsystem (which can use PS post-processing data extracted by deformatter 215 to apply the parametric stereo "PS" tool defined in the MPEG-4 AAC standard), which is coupled and configured to upmix the output of stage 203 to produce a fully decoded upmixed audio signal from APU 210 output.

[0112] Parametric stereo is a coding tool that uses linear downmixing of the left and right channels of a stereo signal and a set of spatial parameters describing the stereo image to represent the stereo signal. Parametric stereo typically uses three types of spatial parameters: (1) inter-channel intensity difference (IID), which describes the intensity difference between channels; (2) inter-channel phase difference (IPD), which describes the phase difference between channels; and (3) inter-channel coherence (ICC), which describes the coherence (or similarity) between channels. Coherence can be measured as the maximum value of cross-correlation that varies with time or phase. These three parameters generally enable high-quality reconstruction of stereo images. However, the IPD parameter only specifies the relative phase difference between the channels of the stereo input signal and does not indicate the distribution of these phase differences on the left and right channels. Therefore, a fourth type of parameter describing a total phase shift or total phase difference (OPD) can be used. In the stereo reconstruction process, the continuous window segments of the received downmixed signal s[n] and the received downmixed decorrelation type d[n], along with spatial parameters, are processed to generate the left (lk(n)) and right (rk(n)) reconstructed signals according to the following equation: H11, H12, H21, and H22 are defined by stereo parameters. Finally, the signals lk(n) and rk(n) are transformed back to the time domain by a frequency-to-time transformation.

[0113] The control data generation subsystem 401 in Figure 5 is coupled and configured to detect at least one property of the encoded audio bitstream to be decoded and to generate eSBR control data (which may be or include any type of eSBR post-data contained in the encoded audio bitstream according to other embodiments of the present invention) in response to at least one result of the detection step. The eSBR control data is verified to level 203 to trigger the application of individual eSBR tools or combinations of eSBR tools and / or control the application of such eSBR tools after detecting a specific property (or combination of properties) of the bitstream. For example, to control the execution of eSBR processing using harmonic transposition, some embodiments of the control data generation subsystem 401 may include: a music detector (e.g., a simplified form of a conventional music detector) for setting the sbrPatchingMode[ch] parameter in response to detecting whether the bitstream indicates music (and confirming the setting parameter to level 203); a transient detector for setting the sbrOversamplingFlag[ch] parameter in response to detecting the presence or absence of transients in the audio content indicated by the bitstream (and confirming the setting parameter to level 203); and / or a pitch detector for setting the sbrPitchInBinsFlag[ch] and sbrPitchInBins[ch] parameters in response to detecting the pitch of the audio content indicated by the bitstream (and confirming the setting parameter to level 203). Other forms of the invention are audio bitstream decoding methods performed by any embodiment of the decoder of the invention described in this and previous paragraphs.

[0114] This invention includes any embodiment of the APU, system, or apparatus of the present invention configured (e.g., programmed) to perform an encoding or decoding method. Other embodiments of the invention include a system or apparatus configured (e.g., programmed) to perform any embodiment of the methods of the present invention, and a computer-readable medium (e.g., an optical disc) storing program code (e.g., in a non-transitory manner) to implement any embodiment of the methods or steps of the present invention. For example, the system of the present invention may be or include a programmable general-purpose processor, digital signal processor, or microprocessor that uses software or firmware to program and / or otherwise configure to perform various operations on data (including any embodiment of the methods or steps of the present invention). This general-purpose processor may be or include a computer system that includes an input device, a memory, and processing circuitry configured to perform an embodiment of the methods (or steps of the present invention) in response to data verified thereto.

[0115] Embodiments of the present invention may be implemented in hardware, firmware, or software, or a combination of both (e.g., as a programmable logic array). Unless otherwise stated, the algorithms or programs contained in part of the present invention are not inherently associated with any particular computer or other device. Specifically, various general-purpose machines may be used with programs written in accordance with the teachings herein, or more specialized devices (e.g., integrated circuits) may be more readily constructed to perform the required method steps. Thus, the present invention may be implemented on one or more computer program implementations on one or more programmable computer systems (e.g., any of the elements of FIG. 1, or encoder 100 (or an element thereof) of FIG. 2, or decoder 200 (or an element thereof) of FIG. 3, or audio processing unit 210 (or an element thereof) of FIG. 4, or decoder 400 (or an element thereof) of FIG. 5), each of which includes at least one processor, at least one data storage system (including volatile and non-volatile memory and / or storage elements), at least one input device or port, and at least one output device or port. The code is applied to input data to perform the functions described herein and to generate output information. The output information is applied to one or more output devices in a known manner.

[0116] Each of these programs can be implemented in any desired computer language (including machine, assembly, or high-level programming, logic, or object-oriented programming languages) to communicate with a computer system. In either case, the language can be a compiled or interpreted language.

[0117] For example, when implemented by a sequence of computer software instructions, the various functions and steps of embodiments of the present invention can be implemented by a sequence of multi-threaded software instructions running in suitable digital signal processing hardware. In this case, the various devices, steps and functions of the embodiments can correspond to parts of the software instructions.

[0118] Each computer program is preferably stored or downloaded to a storage medium or device (e.g., solid-state memory or media, magnetic or optical media) readable by a general-purpose or special-purpose programmable computer to configure and operate the computer when the storage medium or device is read by the computer system to execute the program described herein. The system of the present invention can also be implemented as a computer-readable storage medium configured to (i.e., store) a computer program, wherein such a configured storage medium causes a computer system to operate in a specific and predefined manner to perform the functions described herein.

[0119] Many embodiments of the invention have been described. However, it should be understood that various modifications can be made without departing from the spirit and scope of the invention. Many modifications and variations of the invention can be made in light of the foregoing teachings. For example, to facilitate efficient implementation, phase shifting can be combined with complex QMF analysis and synthesis filter banks. The analysis filter bank is responsible for filtering the time-domain low-frequency signal generated by the core decoder into a complex number of sub-bands (e.g., QMF sub-bands). The synthesis filter bank is responsible for combining the regenerated high-frequency band generated by a selected HFR technique (as indicated by the received sbrPatchingMode parameter) with the decoded low-frequency band to produce a wideband output audio signal. However, a given filter bank implementation operating in a certain sampling rate mode (e.g., normal dual-rate operation or downsampled SBR mode) should not have a phase shift dependent on the bitstream. The QMF bank used in SBR is a complex exponential extension of the theory of cosine modulation filter banks. It has been shown that when using complex exponential modulation to extend the cosine modulation filter bank, the frequency overlap elimination constraint becomes obsolete. Therefore, for the SBR QMF group, the analysis filter hk(n) and the synthesis filter fk(n) can both be defined by the following equations: ,0≤n≤N,0≤k≤M (1) Where p0(n) is a real-valued symmetric or asymmetric prototype filter (usually a low-pass prototype filter), M represents the number of channels, and N is the order of the prototype filter. The number of channels in the analysis filter bank may differ from the number of channels in the synthesis filter bank. For example, the analysis filter bank may have 32 channels and the synthesis filter bank may have 64 channels. When operating the synthesis filter bank in downsampling mode, the synthesis filter bank may have only 32 channels. Since the subband samples from the filter bank are complex values, an additive feasible channel-dependent phase shift step can be added to the analysis filter bank. This additional phase shift needs to be compensated before the synthesis filter bank. Although the phase shift term can theoretically have any value without disrupting the operation of the QMF analysis / synthesis chain, it can also be constrained to certain values ​​for compliance verification. The SBR signal is affected by the choice of phase factor, while the low-pass signal from the core decoder is not. The audio quality of the output signal is unaffected.

[0120] The coefficients p0(n) of the prototype filter can be defined as a length L of 640, as shown in Table 4 below. Table 4 n p0(n) n p0(n) n p0(n) 0 0.0000000000 214 0.0019765601 428 0.0117623832 1 -0.0005525286 215 -0.0032086896 429 0.0163701258 2 -0.0005617692 216 -0.0085711749 430 0.0207997072 3 -0.0004947518 217 -0.0141288827 431 0.0250307561 4 -0.0004875227 218 -0.0198834129 432 0.0290824006 5 -0.0004893791 219 -0.0258227288 433 0.0329583930 6 -0.0005040714 220 -0.0319531274 434 0.0366418116 7 -0.0005226564 221 -0.0382776572 435 0.0401458278 8 -0.0005466565 222 -0.0447806821 436 0.0434768782 9 -0.0005677802 223 -0.0514804176 437 0.0466303305 10 -0.0005870930 224 -0.0583705326 438 0.0495978676 11 -0.0006132747 225 -0.0654409853 439 0.0524093821 12 -0.0006312493 226 -0.0726943300 440 0.0550460034 13 -0.0006540333 227 -0.0801372934 441 0.0575152691 14 -0.0006777690 228 -0.0877547536 442 0.0598166570 15 -0.0006941614 229 -0.0955533352 443 0.0619602779 16 -0.0007157736 230 -0.1035329531 444 0.0639444805 17 -0.0007255043 231 -0.1116826931 445 0.0657690668 18 -0.0007440941 232 -0.1200077984 446 0.0674525021 19 -0.0007490598 233 -0.1285002850 447 0.0689664013 20 -0.0007681371 234 -0.1371551761 448 0.0703533073 21 -0.0007724848 235 -0.1459766491 449 0.0715826364 22 -0.0007834332 236 -0.1549607071 450 0.0726774642 23 -0.0007779869 237 -0.1640958855 451 0.0736406005 24 -0.0007803664 238 -0.1733808172 452 0.0744664394 25 -0.0007801449 239 -0.1828172548 453 0.0751576255 26 -0.0007757977 240 -0.1923966745 454 0.0757305756 27 -0.0007630793 241 -0.2021250176 455 0.0761748321 28 -0.0007530001 242 -0.2119735853 456 0.0765050718 29 -0.0007319357 243 -0.2219652696 457 0.0767204924 30 -0.0007215391 244 -0.2320690870 458 0.0768230011 31 -0.0006917937 245 -0.2423016884 459 0.0768173975 32 -0.0006650415 246 -0.2526480309 460 0.0767093490 33 -0.0006341594 247 -0.2631053299 461 0.0764992170 34 -0.0005946118 248 -0.2736634040 462 0.0761992479 35 -0.0005564576 249 -0.2843214189 463 0.0758008358 36 -0.0005145572 250 -0.2950716717 464 0.0753137336 37 -0.0004606325 251 -0.3059098575 465 0.0747452558 38 -0.0004095121 252 -0.3168278913 466 0.0741003642 39 -0.0003501175 253 -0.3278113727 467 0.0733620255 40 -0.0002896981 254 -0.3388722693 468 0.0725682583 41 -0.0002098337 255 -0.3499914122 469 0.0717002673 42 -0.0001446380 256 0.3611589903 470 0.0707628710 43 -0.0000617334 257 0.3723795546 471 0.0697630244 44 0.0000134949 258 0.3836350013 472 0.0687043828 45 0.0001094383 259 0.3949211761 473 0.0676075985 46 0.0002043017 260 0.4062317676 474 0.0664367512 47 0.0002949531 261 0.4175696896 475 0.0652247106 48 0.0004026540 262 0.4289119920 476 0.0639715898 49 0.0005107388 263 0.4402553754 477 0.0626857808 50 0.0006239376 264 0.4515996535 478 0.0613455171 51 0.0007458025 265 0.4629308085 479 0.0599837480 52 0.0008608443 266 0.4742453214 480 0.0585915683 53 0.0009885988 267 0.4855253091 481 0.0571616450 54 0.0011250155 268 0.4967708254 482 0.0557173648 55 0.0012577884 269 0.5079817500 483 0.0542452768 56 0.0013902494 270 0.5191234970 484 0.0527630746 57 0.0015443219 271 0.5302240895 485 0.0512556155 58 0.0016868083 272 0.5412553448 486 0.0497385755 59 0.0018348265 273 0.5522051258 487 0.0482165720 60 0.0019841140 274 0.5630789140 488 0.0466843027 61 0.0021461583 275 0.5738524131 489 0.0451488405 62 0.0023017254 276 0.5845403235 490 0.0436097542 63 0.0024625616 277 0.5951123086 491 0.0420649094 64 0.0026201758 278 0.6055783538 492 0.0405349170 65 0.0027870464 279 0.6159109932 493 0.0390053679 66 0.0029469447 280 0.6261242695 494 0.0374812850 67 0.0031125420 281 0.6361980107 495 0.0359697560 68 0.0032739613 282 0.6461269695 496 0.0344620948 69 0.0034418874 283 0.6559016302 497 0.0329754081 70 0.0036008268 284 0.6655139880 498 0.0315017608 71 0.0037603922 285 0.6749663190 499 0.0300502657 72 0.0039207432 286 0.6842353293 500 0.0286072173 73 0.0040819753 287 0.6933282376 501 0.0271859429 74 0.0042264269 288 0.7022388719 502 0.0257875847 75 0.0043730719 289 0.7109410426 503 0.0244160992 76 0.0045209852 290 0.7194462634 504 0.0230680169 77 0.0046606460 291 0.7277448900 505 0.0217467550 78 0.0047932560 292 0.7358211758 506 0.0204531793 79 0.0049137603 293 0.7436827863 507 0.0191872431 80 0.0050393022 294 0.7513137456 508 0.0179433381 81 0.0051407353 295 0.7587080760 509 0.0167324712 82 0.0052461166 296 0.7658674865 510 0.0155405553 83 0.0053471681 297 0.7727780881 511 0.0143904666 84 0.0054196775 298 0.7794287519 512 -0.0132718220 85 0.0054876040 299 0.7858353120 513 -0.0121849995 86 0.0055475714 300 0.7919735841 514 -0.0111315548 87 0.0055938023 301 0.7978466413 515 -0.0101150215 88 0.0056220643 302 0.8034485751 516 -0.0091325329 89 0.0056455196 303 0.8087695004 517 -0.0081798233 90 0.0056389199 304 0.8138191270 518 -0.0072615816 91 0.0056266114 305 0.8185776004 519 -0.0063792293 92 0.0055917128 306 0.8230419890 520 -0.0055337211 93 0.0055404363 307 0.8272275347 521 -0.0047222596 94 0.0054753783 308 0.8311038457 522 -0.0039401124 95 0.0053838975 309 0.8346937361 523 -0.0031933778 96 0.0052715758 310 0.8379717337 524 -0.0024826723 97 0.0051382275 311 0.8409541392 525 -0.0018039472 98 0.0049839687 312 0.8436238281 526 -0.0011568135 99 0.0048109469 313 0.8459818469 527 -0.0005464280 100 0.0046039530 314 0.8480315777 528 0.0000276045 101 0.0043801861 315 0.8497805198 529 0.0005832264 102 0.0041251642 316 0.8511971524 530 0.0010902329 103 0.0038456408 317 0.8523047035 531 0.0015784682 104 0.0035401246 318 0.8531020949 532 0.0020274176 105 0.0032091885 319 0.8535720573 533 0.0024508540 106 0.0028446757 320 0.8537385600 534 0.0028446757 107 0.0024508540 321 0.8535720573 535 0.0032091885 108 0.0020274176 322 0.8531020949 536 0.0035401246 109 0.0015784682 323 0.8523047035 537 0.0038456408 110 0.0010902329 324 0.8511971524 538 0.0041251642 111 0.0005832264 325 0.8497805198 539 0.0043801861 112 0.0000276045 326 0.8480315777 540 0.0046039530 113 -0.0005464280 327 0.8459818469 541 0.0048109469 114 -0.0011568135 328 0.8436238281 542 0.0049839687 115 -0.0018039472 329 0.8409541392 543 0.0051382275 116 -0.0024826723 330 0.8379717337 544 0.0052715758 117 -0.0031933778 331 0.8346937361 545 0.0053838975 118 -0.0039401124 332 0.8311038457 546 0.0054753783 119 -0.0047222596 333 0.8272275347 547 0.0055404363 120 -0.0055337211 334 0.8230419890 548 0.0055917128 121 -0.0063792293 335 0.8185776004 549 0.0056266114 122 -0.0072615816 336 0.8138191270 550 0.0056389199 123 -0.0081798233 337 0.8087695004 551 0.0056455196 124 -0.0091325329 338 0.8034485751 552 0.0056220643 125 -0.0101150215 339 0.7978466413 553 0.0055938023 126 -0.0111315548 340 0.7919735841 554 0.0055475714 127 -0.0121849995 341 0.7858353120 555 0.0054876040 128 0.0132718220 342 0.7794287519 556 0.0054196775 129 0.0143904666 343 0.7727780881 557 0.0053471681 130 0.0155405553 344 0.7658674865 558 0.0052461166 131 0.0167324712 345 0.7587080760 559 0.0051407353 132 0.0179433381 346 0.7513137456 560 0.0050393022 133 0.0191872431 347 0.7436827863 561 0.0049137603 134 0.0204531793 348 0.7358211758 562 0.0047932560 135 0.0217467550 349 0.7277448900 563 0.0046606460 136 0.0230680169 350 0.7194462634 564 0.0045209852 137 0.0244160992 351 0.7109410426 565 0.0043730719 138 0.0257875847 352 0.7022388719 566 0.0042264269 139 0.0271859429 353 0.6933282376 567 0.0040819753 140 0.0286072173 354 0.6842353293 568 0.0039207432 141 0.0300502657 355 0.6749663190 569 0.0037603922 142 0.0315017608 356 0.6655139880 570 0.0036008268 143 0.0329754081 357 0.6559016302 571 0.0034418874 144 0.0344620948 358 0.6461269695 572 0.0032739613 145 0.0359697560 359 0.6361980107 573 0.0031125420 146 0.0374812850 360 0.6261242695 574 0.0029469447 147 0.0390053679 361 0.6159109932 575 0.0027870464 148 0.0405349170 362 0.6055783538 576 0.0026201758 149 0.0420649094 363 0.5951123086 577 0.0024625616 150 0.0436097542 364 0.5845403235 578 0.0023017254 151 0.0451488405 365 0.5738524131 579 0.0021461583 152 0.0466843027 366 0.5630789140 580 0.0019841140 153 0.0482165720 367 0.5522051258 581 0.0018348265 154 0.0497385755 368 0.5412553448 582 0.0016868083 155 0.0512556155 369 0.5302240895 583 0.0015443219 156 0.0527630746 370 0.5191234970 584 0.0013902494 157 0.0542452768 371 0.5079817500 585 0.0012577884 158 0.0557173648 372 0.4967708254 586 0.0011250155 159 0.0571616450 373 0.4855253091 587 0.0009885988 160 0.0585915683 374 0.4742453214 588 0.0008608443 161 0.0599837480 375 0.4629308085 589 0.0007458025 162 0.0613455171 376 0.4515996535 590 0.0006239376 163 0.0626857808 377 0.4402553754 591 0.0005107388 164 0.0639715898 378 0.4289119920 592 0.0004026540 165 0.0652247106 379 0.4175696896 593 0.0002949531 166 0.0664367512 380 0.4062317676 594 0.0002043017 167 0.0676075985 381 0.3949211761 595 0.0001094383 168 0.0687043828 382 0.3836350013 596 0.0000134949 169 0.0697630244 383 0.3723795546 597 -0.0000617334 170 0.0707628710 384 -0.3611589903 598 -0.0001446380 171 0.0717002673 385 -0.3499914122 599 -0.0002098337 172 0.0725682583 386 -0.3388722693 600 -0.0002896981 173 0.0733620255 387 -0.3278113727 601 -0.0003501175 174 0.0741003642 388 -0.3168278913 602 -0.0004095121 175 0.0747452558 389 -0.3059098575 603 -0.0004606325 176 0.0753137336 390 -0.2950716717 604 -0.0005145572 177 0.0758008358 391 -0.2843214189 605 -0.0005564576 178 0.0761992479 392 -0.2736634040 606 -0.0005946118 179 0.0764992170 393 -0.2631053299 607 -0.0006341594 180 0.0767093490 394 -0.2526480309 608 -0.0006650415 181 0.0768173975 395 -0.2423016884 609 -0.0006917937 182 0.0768230011 396 -0.2320690870 610 -0.0007215391 183 0.0767204924 397 -0.2219652696 611 -0.0007319357 184 0.0765050718 398 -0.2119735853 612 -0.0007530001 185 0.0761748321 399 -0.2021250176 613 -0.0007630793 186 0.0757305756 400 -0.1923966745 614 -0.0007757977 187 0.0751576255 401 -0.1828172548 615 -0.0007801449 188 0.0744664394 402 -0.1733808172 616 -0.0007803664 189 0.0736406005 403 -0.1640958855 617 -0.0007779869 190 0.0726774642 404 -0.1549607071 618 -0.0007834332 191 0.0715826364 405 -0.1459766491 619 -0.0007724848 192 0.0703533073 406 -0.1371551761 620 -0.0007681371 193 0.0689664013 407 -0.1285002850 621 -0.0007490598 194 0.0674525021 408 -0.1200077984 622 -0.0007440941 195 0.0657690668 409 -0.1116826931 623 -0.0007255043 196 0.0639444805 410 -0.1035329531 624 -0.0007157736 197 0.0619602779 411 -0.0955533352 625 -0.0006941614 198 0.0598166570 412 -0.0877547536 626 -0.0006777690 199 0.0575152691 413 -0.0801372934 627 -0.0006540333 200 0.0550460034 414 -0.0726943300 628 -0.0006312493 201 0.0524093821 415 -0.0654409853 629 -0.0006132747 202 0.0495978676 416 -0.0583705326 630 -0.0005870930 203 0.0466303305 417 -0.0514804176 631 -0.0005677802 204 0.0434768782 418 -0.0447806821 632 -0.0005466565 205 0.0401458278 419 -0.0382776572 633 -0.0005226564 206 0.0366418116 420 -0.0319531274 634 -0.0005040714 207 0.0329583930 421 -0.0258227288 635 -0.0004893791 208 0.0290824006 422 -0.0198834129 636 -0.0004875227 209 0.0250307561 423 -0.0141288827 637 -0.0004947518 210 0.0207997072 424 -0.0085711749 638 -0.0005617692 211 0.0163701258 425 -0.0032086896 639 -0.0005525280 212 0.0117623832 426 0.0019765601 213 0.0069636862 427 0.0069636862 The prototype filter p0(n) can also be derived from Table 4 by one or more mathematical operations such as rounding, subsampling, interpolation and sampling.

[0121] Although the tuning of SBR-related control information generally does not depend on the transpose details (as previously discussed), in some embodiments, certain elements of the control data can be cascaded in the eSBR extension area (bs_extension_id==EXTENSION_ID_ESBR) to improve the quality of the regenerated signal. Some cascaded elements may include noise limit data (e.g., a parameter indicating the direction (frequency or time direction) of differential encoding for each noise limit), inverse filtering data (e.g., a parameter indicating an inverse filtering mode selected from no inverse filtering, a low inverse filtering level, a moderate inverse filtering level, and a strong inverse filtering level), and missing harmonic data (e.g., a parameter indicating whether a sine wave should be added to a specific frequency band of the regenerated high-frequency band). All of these elements rely on a synthetic analog of the transposer of the decoder executed in the encoder and thus can improve the quality of the regenerated signal after appropriate tuning according to the selected transposer.

[0122] Specifically, in some embodiments, missing harmonic and inverse filter control data (along with other bitstream parameters in Table 3) are transmitted in the eSBR extended region and tuned according to the eSBR's harmonic transposer. The additional bit rate required to transmit these two types of post-processing data from the eSBR's harmonic converter is relatively low. Therefore, transmitting the tuning missing harmonic and / or inverse filter control data in the eSBR extended region will improve the quality of the audio generated by the transposer while only slightly affecting the bit rate. To ensure backtracking compatibility with legacy decoders, implicit or explicit transmission can also be used in the bitstream to transmit the parameters tuned according to the SBR's spectrum shift operation as part of the SBR control data.

[0123] The complexity of the decoder with SBR enhancement described in this application must be limited to avoid significantly increasing the overall computational complexity of the implementation. Preferably, when using the eSBR tool, the PCU (MOP) of the SBR object type is equal to or less than 4.5, and the RCU of the SBR object type is equal to or less than 3. Approximate processing power is given in processor complexity units (PCU) (specified by an integer number of MOPS). Approximate RAM usage is given in RAM complexity units (RCU) (specified by an integer number of kWords (1000 words)). The RCU number does not include working buffers that can be shared between different objects and / or channels. Furthermore, the PCU is proportional to the sampling frequency. PCU values ​​are given in MOPS per channel (millions of operations per second), and RCU values ​​are given in kilowords per channel.

[0124] Special attention needs to be paid to compressed data, such as HE-AAC encoded audio that can be decoded by different decoder configurations. In this case, decoding can be performed in a backtracking compatible mode (AAC only) and an enhanced mode (AAC+SBR). If the compressed data allows both backtracking compatible and enhanced decoding, and if the decoder operates in enhanced mode, causing it to use a post-processor with some additional delay (e.g., an SBR post-processor in HE-AAC), then it must be ensured that this additional time delay caused by the backtracking compatible mode is taken into account when presenting the combining unit, as described by a corresponding value n. To ensure proper handling of the combining timestamp (so that the audio is synchronized with other media), when the decoder operating mode includes the SBR enhancement described in this application (including eSBR), the additional delay introduced by post-processing, given the number of samples (per audio channel) at the output sampling rate, is 3010. Therefore, for an audio combining unit, when the decoder operating mode includes the SBR enhancement described in this application, the combining time is applied to the 3011th audio sample within the combining unit.

[0125] SBR enhancement should be enabled to improve the subjective quality of audio content with harmonic frequency structures and strong tonal characteristics, especially at low bit rates. The value of the corresponding bitstream element (i.e., esbr_data()) controlling these tools can be determined in the encoder by applying a signal dependency classification mechanism.

[0126] Generally speaking, the SBR patching mode (SBRPatchingMode==0) is more suitable for encoding music signals at very low bit rates, where the audio bandwidth of the core codec is greatly limited. This is especially true when such signals contain a significant harmonic structure. Conversely, the conventional SBR patching mode is more suitable for speech and mixed signals because it provides better preservation of one of the temporal structures of speech.

[0127] To improve the performance of the MPEG-4 SBR transposer, a preprocessing step (bs_sbr_preprocessing==1) can be initiated to avoid introducing spectral discontinuities into the signal entering the subsequent envelope adjuster. The tool's operation is beneficial for displaying the coarse spectral envelope of low-frequency band signals used for high-frequency reconstruction, revealing signal types with large level variations.

[0128] To improve the transient response of harmonic SBR patching (sbrPatchingMode==0), adaptive frequency domain oversampling (sbrOversamplingFlag==1) can be applied. Since adaptive frequency domain oversampling increases the computational complexity of the transposer, but only benefits frames containing transients, the use of this tool is controlled by bitstream elements, which are transmitted once per frame and per independent SBR channel.

[0129] Typical bitrate settings for HE-AACv2 with SBR enhancement (i.e., harmonic transposer with eSBR enabled) correspond to 20 kbps to 32 kbps of stereo audio content at sampling rates of 44.1 kHz or 48 kHz. The relative subjective quality gain of SBR enhancement increases towards lower bitrate boundaries, and a properly configured encoder allows this range to be extended to even lower bitrates. The bitrates provided above are recommendations only and may be applicable to specific service requirements.

[0130] In the proposed enhanced SBR mode, one of the decoders typically needs to be able to switch between legacy SBR patch and enhanced SBR patch. Therefore, depending on the decoder settings, a delay as long as the duration of a core audio message frame can be introduced. Generally, the delays for legacy SBR patch and enhanced SBR patch will be similar.

[0131] It should be understood that the invention may be practiced in ways other than those specifically described herein, within the scope of the appended claims. Any element symbols contained in the following claims are for illustrative purposes only and should not be used to interpret or limit the scope of the claims.

[0132] Various aspects of the invention can be understood from the following exemplary embodiments (EEE):

[0133] EEE 1. A method for performing high-frequency reconstruction of an audio signal, the method comprising: Receive an encoded audio bitstream, the encoded audio bitstream containing audio data representing a low-frequency band portion of the audio signal and high-frequency reconstruction post-processing data; Decode the audio data to generate a decoded low-frequency audio signal; The high-frequency reconstruction post-processing data is extracted from the encoded audio bitstream. The high-frequency reconstruction post-processing data includes operating parameters of a high-frequency reconstruction procedure. These operating parameters include a patch mode parameter located in a backtracking compatible extension region of the encoded audio bitstream. A first value of the patch mode parameter indicates a spectral shift, and a second value of the patch mode parameter indicates harmonic transposition by frequency extension via a phase vocoder. The decoded low-frequency audio signal is filtered to generate a filtered low-frequency audio signal; Using the filtered low-frequency audio signal and the high-frequency reconstructed post-processing data, a high-frequency portion of the audio signal is regenerated. If the repair mode parameter is the first value, the regeneration includes spectral shifting; and if the repair mode parameter is the second value, the regeneration includes harmonic transposition via frequency extension using a phase vocoder. The filtered low-frequency audio signal is combined with the regenerated high-frequency portion to form a wideband audio signal. The filtering, regeneration, and combination are performed as a post-processing operation with a delay of 3010 samples per audio channel or less, and the spectral shift includes maintaining a ratio between the tone component and the noise-like component by means of adaptive inverse filtering.

[0134] EEE 2. The method of EEE 1, wherein the encoded audio bitstream further includes a padding element having an identifier indicating the start of one of the padding elements and padding data following the identifier, wherein the padding data includes the backtracking compatible extension area.

[0135] EEE 3. The method of EEE 2, wherein the identifier is transmitted first as a 3-bit unsigned integer with a value of 0×6 and the most significant bit first.

[0136] EEE 4. The method of EEE 2 or EEE 3, wherein the padding data includes an extended payload comprising spectrum band copy extension data, and the extended payload is identified by a 4-bit unsigned integer having a value of either "1101" or "1110" that was transmitted first, as appropriate. The extended data for this spectrum band replication includes: Select a spectrum band copy header. The spectrum band copy data, which is located after this header, and A spectrum band replication extension element, which is located after the spectrum band replication data, wherein the flag is contained within the spectrum band replication extension element.

[0137] EEE 5. The method of any one of EEE 1 to 4, wherein the high-frequency reconstruction post-processing data includes an envelope scaling factor, a noise threshold scaling factor, time / frequency grid information, or a parameter indicating an interleaving frequency.

[0138] EEE 6. The method of any one of EEE 1 to 5, wherein the retrospective compatibility extension region further includes a flag indicating whether additional preprocessing is used to avoid shape discontinuities in the spectral envelope of one of the high-frequency band portions when the patch mode parameter is equal to the first value, wherein a first value of the flag enables the additional preprocessing and a second value of the flag disables the additional preprocessing.

[0139] EEE 7. The method of EEE 6, wherein the additional preprocessing includes using a linear prediction filter coefficient to calculate a pregain curve.

[0140] EEE 8. The method of any one of EEE 1 to 5, wherein the backtracking compatible extended capacity further includes a flag indicating whether signal adaptive frequency domain oversampling is applied when the patch mode parameter is equal to the second value, wherein a first value of the flag enables the signal adaptive frequency domain oversampling and a second value of the flag disables the signal adaptive frequency domain oversampling.

[0141] EEE 9. The method of EEE 8, wherein the adaptive frequency domain oversampling of the signal is applied only to a frame containing a transient state.

[0142] EEE 10. The method of any of the aforementioned EEEs, wherein the harmonic transpose is performed by frequency extension of the phase vocoder with an estimated complexity equal to or less than 4.5 million operations per second and 3,000 words of memory.

[0143] EEE 11. A non-transitory computer-readable medium containing instructions that, when executed by a processor, perform a method as described in any of EEE 1 to 10.

[0144] EEE 12. A computer program product having instructions which, when executed by a computing device or system, cause the computing device or system to perform a method as described in any one of EEE 1 to 10.

[0145] EEE 13. An audio processing unit for performing high-frequency reconstruction of an audio signal, the audio processing unit comprising: An input interface for receiving an encoded audio bitstream, the encoded audio bitstream containing audio data representing a low-frequency band portion of the audio signal and high-frequency reconstruction post-processing data; A core audio decoder is used to decode the audio data to generate a decoded low-frequency audio signal; A deformatter is used to extract high-frequency reconstruction post-processing data from the encoded audio bitstream. The high-frequency reconstruction post-processing data includes operating parameters for a high-frequency reconstruction procedure. These operating parameters include a patch mode parameter located within a backtracking compatible extension region of the encoded audio bitstream. A first value of the patch mode parameter indicates a spectral shift, and a second value of the patch mode parameter indicates harmonic transposition via frequency extension by a phase vocoder. An analysis filter bank is used to filter the decoded low-frequency audio signal to generate a filtered low-frequency audio signal; A high-frequency regenerator is used to reconstruct a high-frequency band portion of the audio signal using a filtered low-frequency band audio signal and the high-frequency reconstruction post-processing data, wherein if the repair mode parameter is a first value, the reconstruction includes a spectral shift, and if the repair mode parameter is a second value, the reconstruction includes harmonic transposition by frequency extension of a phase vocoder; and A synthesizing filter bank is used to combine the filtered low-frequency audio signal with the regenerated high-frequency portion to form a wideband audio signal. The analysis filter bank, the high-frequency regenerator, and the synthesis filter bank are executed in a postprocessor with a delay of 3010 samples or less per audio channel, and the spectral shift includes maintaining a ratio between the tone component and the noise-like component by means of adaptive inverse filtering.

[0146] EEE 14. The audio processing unit of EEE 13, wherein the harmonic transpose is performed by frequency extension of the phase vocoder with an estimated complexity of 4.5 million operations per second or less and 3,000 words of memory.

[0147] 1: Encoder 2: Transmission Subsystem 3: Decoder 4: Post-processing unit 100: Encoder 105: Encoder 106: Metadata Generation Stage / Metadata Generator 107: Filler / Formatter Level 109: Buffer Memory 200: Decoder 201: Buffer memory / buffer 202: Audio Decoding Subsystem / Core Decoding Subsystem 203: Enhanced Spectrum Band Replication (eSBR) Processing Level 204: Control bit generation stage / control bit generator 205: Bitstream payload deformatter / parser 210: Audio Processing Unit (APU) 213: Spectrum Band Replication (SBR) Processing Level 215: Bitstream payload deformatter 300: Post-processor 301: Buffer / Buffered Memory 400: Decoder 401: eSBR Control Data Generation Subsystem 500: APU ID1: Identifier ID2: Identifier

Claims

1. A method for performing high-frequency reconstruction of an audio signal, the method comprising: Receive an encoded audio bitstream, the encoded audio bitstream comprising audio data representing a low-frequency band portion of the audio signal and high-frequency reconstruction post-processing data, wherein the encoded audio bitstream further comprises a padding element having an identifier indicating the start of the padding element and padding data following the identifier; decode the audio data to generate a decoded low-frequency band audio signal; extract the high-frequency reconstruction post-processing data from the encoded audio bitstream, the high-frequency reconstruction post-processing data comprising operating parameters for a high-frequency reconstruction procedure, the operating parameters comprising a patch mode parameter located in a backtracking compatible extension container of the encoded audio bitstream, wherein a first value of the patch mode parameter indicates spectral translation and a second value of the patch mode parameter indicates harmonic transposition by frequency spreading via a phase vocoder, wherein the padding data comprises the backtracking compatible extension container; The decoded low-frequency audio signal is filtered to generate a filtered low-frequency audio signal; and a high-frequency band portion of the audio signal is regenerated using the filtered low-frequency audio signal and the high-frequency reconstruction post-processing data, wherein if the repair mode parameter is the first value, the regeneration includes spectral shifting, and if the repair mode parameter is the second value, the regeneration includes harmonic transposition by frequency extension of a phase vocoder; wherein the filtering and the regeneration are performed as a post-processing operation with a delay of 3010 samples per audio channel, and wherein the spectral shifting includes maintaining a ratio between the tone component and the noise-like component by adaptive inverse filtering.

2. The method of claim 1, wherein the backtracking compatibility extension region further includes a flag indicating whether additional preprocessing is used to avoid shape discontinuities in the spectral envelope of one of the high-frequency band portions when the patch mode parameter is equal to the first value, wherein a first value of the flag enables the additional preprocessing and a second value of the flag disables the additional preprocessing.

3. The method of request item 2, wherein the additional preprocessing includes using a linear prediction filter coefficient to calculate a pregain curve.

4. The method of claim 1, wherein the backtracking compatible extension region further includes a flag indicating whether to apply signal adaptive frequency domain oversampling when the patch mode parameter is equal to the second value, wherein a first value of the flag enables the signal adaptive frequency domain oversampling and a second value of the flag disables the signal adaptive frequency domain oversampling.

5. The method of claim 4, wherein the adaptive frequency domain oversampling of the signal is applied only to a frame containing a transient state.

6. The method of claim 1, wherein the harmonic transpose is performed by frequency extension of the phase vocoder with an estimated complexity of 4.5 million operations per second or less and 3,000 words of memory or less.

7. A non-transitory computer-readable medium containing instructions that, when executed by a processor, perform the method as requested in claim 1.

8. A computer program product stored in a non-transitory computer-readable medium having instructions that, when executed by a computing device or system, cause the computing device or system to perform the method as claimed in claim 1.

9. An audio processing unit for performing high-frequency reconstruction of an audio signal, the audio processing unit comprising: An input interface for receiving an encoded audio bitstream, the encoded audio bitstream including audio data representing a low-frequency band portion of the audio signal and high-frequency reconstruction post-processing data, wherein the encoded audio bitstream further includes a padding element having an identifier indicating the start of the padding element and padding data following the identifier; a core audio decoder for decoding the audio data to generate a decoded low-frequency audio signal; a deformatter for extracting the high-frequency reconstruction post-processing data from the encoded audio bitstream, the high-frequency reconstruction post-processing data including operating parameters for a high-frequency reconstruction procedure, the operating parameters including a patch mode parameter located in a backtracking compatible extension region of the encoded audio bitstream, wherein a first value of the patch mode parameter indicates a spectral shift and a second value of the patch mode parameter indicates a harmonic transpose by frequency extension of a phase vocoder, wherein the padding data includes the backtracking compatible extension region; An analysis filter bank for filtering the decoded low-frequency band audio signal to generate a filtered low-frequency band audio signal; and a high-frequency regenerator for reconstructing a high-frequency band portion of the audio signal using the filtered low-frequency band audio signal and the high-frequency reconstruction post-processor data, wherein if the patching mode parameter is the first value, the reconstruction includes a spectral shift, and if the patching mode parameter is the second value, the reconstruction includes harmonic transposition by frequency extension of a phase vocoder; wherein the analysis filter bank and the high-frequency regenerator are executed in a post-processor having a delay of 3010 samples per audio channel, and wherein the spectral shift includes maintaining a ratio between the tone component and the pseudo-noise component by adaptive inverse filtering.

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