ES2948839T3 - Backwards compatible integration of high frequency reconstruction techniques for audio signals - Google Patents

Abstract

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded low band audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded low band audio signal with a bank of analysis filters to generate a filtered low band audio signal. The method also includes extracting a flag indicating whether spectral translation or harmonic transposition is to be performed on the audio data and regenerating a high-band portion of the audio signal using the filtered low-band audio signal and the reconstruction metadata. high frequency according to the indicator. (Automatic translation with Google Translate, without legal value)

Description

DESCRIPTION

Backwards compatible integration of high frequency reconstruction techniques for audio signals Cross reference to related request

This application is a European divisional application of European patent application EP 21164481.0 (reference: D17806EP02), the EPO Form 1001 of which was filed on March 24, 2021.

Technical field

The embodiments pertain to a method and an audio processing unit for performing high frequency reconstruction of an audio signal.

Background of the invention

A typical audio bitstream includes both audio data (e.g., encoded audio data) indicative of one or more channels of audio content, and metadata indicative of at least one characteristic of the audio data or audio content. . A well-known format for generating an encoded 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 denotes "advanced audio coding" and HE-AAC denotes "high efficiency advanced audio coding".

The MPEG-4 AAC standard defines several audio profiles, which determine which encoding objects and tools are present in a compatible encoder or decoder. Three of these 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 includes neither the Spectral Band Replication ("SBR") object type nor 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.

The SBR object type contains the Spectral Band Replication tool, which is an important high-frequency reconstruction ("HFR") coding tool that significantly improves the compression efficiency of perceptual audio codecs. SBR reconstructs the high-frequency components of an audio signal at the receiver side (e.g. at the decoder). Therefore, the encoder only needs to encode and transmit low-frequency components, allowing for much higher audio quality at low data rates. SBR is based on the replication of harmonic sequences, previously truncated to reduce the data rate, from the available bandwidth limited signal and control data obtained from the encoder. The relationship between tonal and noise-like components is maintained by adaptive inverse filtering, as well as the optional addition of noise and sine. In the MPEG-4 AAC standard, the SBR tool performs spectral patching (also called linear translation or spectral translation), where several consecutive subbands of the quadrature mirror filter (QMF) are copied (or "patched") from a portion transmitted low-band portion of an audio signal to a high-band portion of the audio signal, which is generated in the decoder.

Spectral patching or linear translation may not be ideal for certain types of audio, such as musical content with relatively low crossover frequencies. Therefore, techniques are needed to improve spectral band replication.

WO 2016/149015 A1 describes an audio processing unit that includes a buffer, a bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a padding element that begins with an identifier followed by padding data. The padding data includes at least one flag that identifies whether enhanced spectral band replication (eSBR) processing is to be performed on the audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Summary of the invention

The invention is defined by the independent claims. Specific embodiments are defined in the dependent claims.

Brief description of embodiments of the invention

A first class of embodiments relates to a method according to claim 1.

A second class of embodiments relates to an audio processing unit according to claim 7.

Brief description of the drawings

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

Figure 2 is a block diagram of an encoder which is an embodiment of the audio processing unit of the invention.

Figure 3 is a block diagram of a system that includes a decoder that is an embodiment of the audio processing unit of the invention and, optionally, also a postprocessor coupled thereto.

Figure 4 is a block diagram of a decoder which is an embodiment of the audio processing unit of the invention.

Figure 5 is a block diagram of a decoder which is another embodiment of the audio processing unit of the invention.

Figure 6 is a block diagram of another embodiment of the audio processing unit of the invention. Figure 7 is a block diagram of an MPEG-4 AAC bitstream, including the segments into which it is divided.

Nomenclature and notation

Throughout this disclosure, including the claims, the term perform an operation "on" a signal or data (e.g., filter, scale, transform or apply gain to the signal or data) is used in a broad sense to denote performing the operation directly on the signal or data, or on a processed version of the signal or data (for example, on a version of the signal that has been subjected to preliminary filtering or preprocessing before performing the operation on the same).

Throughout this disclosure, including the claims, the term "audio processing unit" or "audio processor" is used in a broad sense, to indicate a system, device or apparatus, configured to process audio data. Examples of audio processing units include, but are not limited to, encoders, transcoders, decoders, codecs, preprocessing systems, postprocessing systems, and bitstream processing systems (sometimes called bitstream processing tools). Virtually all consumer electronics products, such as mobile phones, televisions, laptops and tablets, contain an audio processing unit or audio processor.

Throughout this disclosure, including the claims, the term "couples" or "coupled" is used in a broad sense to mean a direct or indirect connection. Therefore, if a first device is coupled to a second device, that connection can be through a direct connection or through an indirect connection through other devices and connections. Furthermore, components that are integrated into or with other components are also coupled to each other.

Detailed description of embodiments of the invention

The MPEG-4 AAC standard contemplates that an encoded MPEG-4 AAC bitstream includes metadata indicative of each type of high frequency reconstruction ("HFR") processing that will be applied (if any) by a decoder to decode the audio content of the bitstream, and/or that controls such HFR processing, and/or is indicative of at least one characteristic or parameter of at least one HFR tool that will be used to decode the audio content of the bitstream. The term "SBR metadata" is used herein to denote such metadata that is described or referenced in the MPEG-4 AAC standard for use with spectral band replication ("SBR"). As one skilled in the art will appreciate, SBR is a form of HFR.

SBR is preferably used as a dual-rate system, with the underlying codec operating at half the original sampling rate, while SBR operates at the original sampling rate. The SBR encoder works in parallel with the underlying core codec, albeit at a higher sample rate. Although SBR is mainly a post-process in the decoder, important parameters are extracted in the encoder in order to ensure the most accurate high-frequency reconstruction in the decoder. The encoder estimates the spectral envelope of the SBR range for a time and frequency range/resolution appropriate for the characteristics of the current input signal segments. The spectral envelope is estimated by complex QMF analysis and subsequent energy calculation. The time and frequency resolutions of the envelopes Spectral frequencies can be chosen with a high level of freedom, in order to ensure the most appropriate time-frequency resolution for the given input segment. The envelope estimation must take into account that a transient in the original, mainly located in the high frequency region (e.g., a high zone), will be present to a lesser extent in the high band generated by SBR before adjustment of the envelope. envelope, since the high band in the decoder is based on the low band where the transient is much less pronounced compared to the high band. This aspect imposes different requirements for the time-frequency resolution of the spectral envelope data, compared to ordinary spectral envelope estimation as used in other audio coding algorithms.

In addition to the spectral envelope, several additional parameters are extracted that represent the spectral characteristics of the input signal for different time and frequency regions. Since the encoder naturally has access to the original signal, as well as information about how the SBR unit in the decoder will create the high band, given the specific set of control parameters, it is possible for the system to handle situations where the high band The low band constitutes a series of strong harmonics and the high band, which is to be recreated, mainly constitutes random signal components, as well as situations in which strong tonal components are present in the original high band without counterparts in the low band, over the which is based on the high band region. Additionally, the SBR encoder works closely with the underlying core codec to evaluate what frequency range should be covered by SBR at any given time. SBR data is efficiently encoded before transmission by exploiting entropy coding, as well as the channel dependencies of the control data, in the case of stereo signals.

Typically, control parameter extraction algorithms must be carefully tuned to the underlying codec at a given bitrate and sample rate. This is due to the fact that a lower bit rate generally implies a larger SBR range compared to a high bit rate, and different sampling rates correspond to different time resolutions of the SBR frames.

An SBR decoder typically includes several different parts. It comprises a bitstream decoding module, a high-frequency reconstruction (HFR) module, an additional high-frequency component module, and an envelope adjustment module. The system is based on a complex value QMF filter bank (for high quality SBR) or a real value QMF filter bank (for low power SBR). Embodiments of the invention are applicable to both high quality SBR and low power SBR. In the bitstream extraction module, control data is read from the bitstream and decoded. The time-frequency grid is obtained for the current frame, before reading the data from the bitstream envelope. The underlying core decoder decodes the audio signal of the current frame (albeit at the lowest sampling rate) to produce time domain audio samples. The resulting frame of audio data is used for high frequency reconstruction by the HFR module. The decoded low band signal is then analyzed using a QMF filter bank. High frequency reconstruction and envelope adjustment are subsequently performed on the subband samples of the QMF filter bank. The high frequencies are reconstructed from the low band flexibly, based on the given control parameters. Additionally, the reconstructed high band is adaptively filtered on a subband channel basis according to the control data to ensure appropriate spectral characteristics of the given time/frequency region.

The top level of an MPEG-4 AAC bitstream is a sequence of data blocks ("raw_data_block" elements), each of which is a data segment (herein referred to as a "block") containing audio (typically over a time period of 1024 or 960 samples) and related information and/or other data. Herein, the term "block" is used to denote a segment of an MPEG-4 AAC bitstream comprising audio data (and corresponding metadata and, optionally, also other related data) that determines or is indicative of one "raw_data_block" element (but not more than one).

Each block of an MPEG-4 AAC bitstream can include several syntactic elements (each of which is also embodied in the bitstream as a data segment). Seven types of such syntactic elements are defined in the MPEG-4 AAC standard. Each syntax element is identified by a different value of the "id_syn_ele" data element. Examples of syntactic elements include a "single_channeLelement()", a "channeLpair_element()", and a "filLelement()". A single-channel element is a container that includes audio data from a single audio channel (a signal monophonic audio). A channel pair element includes audio data from two audio channels (that is, a stereo audio signal).

A padding element is a container of information that includes an identifier (for example, the value of the "id_syn_ele" element above) followed by data, which is called "padding data." Padding elements have historically been used to adjust the instantaneous bitrate of bitstreams to be transmitted over a constant rate channel. By adding the appropriate amount of padding data to each block, a constant data rate can be achieved.

In accordance with embodiments of the invention, the padding data may include one or more extension payloads that extend the type of data (e.g., metadata) that can be transmitted in a bitstream. A A decoder that receives bitstreams with padding data containing a new type of data may optionally be used by a device that receives the bitstream (e.g., a decoder) to extend the functionality of the device. Therefore, as one skilled in the art can appreciate, padding elements are a special type of data structure and are different from data structures that are typically used to transmit audio data (for example, audio payloads that contain channel data).

In some embodiments of the invention, the identifier used to identify a filler element may consist of a three-bit unsigned integer transmitted most significant bit first ("uimsbf") having a value of 0x6. In a block, multiple instances of the same type of syntactic element (for example, multiple filler elements) can occur.

Another standard for encoding audio bitstreams is the MPEG Unified Speech and Audio Coding (USAC) standard (ISO/IEC 23003-3:2012). The USAC MPEG standard describes the encoding and decoding of audio content using spectral band replication processing (including SBR processing as described in the MPEG-4 AAC standard, and also includes other enhanced forms of spectral band replication processing) . This processing applies spectral band replication tools (sometimes referred to herein as "enhanced SBR tools" or "eSBR tools") from an expanded and improved version of the set of SBR tools described in the MPEG-4 AAC standard. Therefore, eSBR (as defined in the USAC standard) is an enhancement of SBR (as defined in the MPEG-4 AAC standard).

Herein, the term "enhanced SBR processing" (or "eSBR processing") is used to denote spectral band replication processing using at least one eSBR tool (e.g., at least one eSBR tool described or mentioned in the USAC MPEG standard) that is not described or mentioned in the MPEG-4 AAC standard. Examples of such eSBR tools are harmonic transposition and additional QMF patching preprocessing or "pre-flattening".

An integer-order harmonic transponder T maps a sinusoid with frequency w to a sinusoid with frequency Tw, while preserving signal duration. Typically, three orders, T = 2, 3, 4 are used in sequence to produce each part of the desired output frequency range using the smallest possible transposition order. If output above the fourth order transposition range is required, it can be generated by frequency changes. When possible, quasi-critically sampled baseband time domains are created so that processing minimizes computational complexity.

The harmonic transponder can be based on QMF or DFT. When the QMF-based harmonic transponder is used, the extension of the time domain signal bandwidth of the core encoder is carried out entirely in the QMF domain, using a modified phase vocoder structure, performing decimation followed by time stretching for each QMF subband. Transposition using various transposition factors (e.g., T = 2, 3, 4) is carried out in a common QMF analysis/synthesis transformation step. Since the QMF-based harmonic transponder does not feature signal adaptive frequency domain oversampling, the corresponding flag in the bitstream (sbrOversamplingFlag [ch]) can be ignored.

When the DFT-based harmonic transponder is used, the factor 3 and 4 transponders (3rd and 4th order transponders) are preferably integrated into the factor 2 transponder (2nd order transponder) by means of interpolation to reduce complexity . For each frame (corresponding to core encoder coreCoderFrameLength samples), the nominal "full-size" transponder transformation size is first determined by the signal's adaptive frequency-domain oversampling flag (sbrOversamplingFlag [ch]) in the bit stream.

When sbrPatchingMode == 1, indicating that linear transposition is to be used to generate the high band, an additional step can be introduced to avoid discontinuities in the shape of the spectral envelope of the high frequency signal that is fed into the fitter rear envelope. This improves the performance of the next stage of envelope adjustment, resulting in a high-band signal that is perceived as more stable. The operation of additional preprocessing is beneficial for signal types where the thick spectral envelope of the low-band signal used for high-frequency reconstruction shows large level variations. However, the value of the bitstream element can be determined in the encoder by applying any type of signal-dependent classification. Additional preprocessing is preferably enabled via a one-bit bitstream element, bs_sbr_preprocessing. When bs_sbr_preprocessing is set to one, additional processing is enabled. When bs_sbr_preprocessing is set to zero, additional preprocessing is disabled. Preferred additional processing uses a preGain curve that uses the high frequency generator to scale the low band, XLow, for each patch. For example, the preGain curve can be calculated according to:

preGain ( k) - io ( meanNra~íowEnvSloPeW)/20io < k < k 0

where ko is the first QMF subband in the master frequency band table and lowEnvSlope is calculated using a function that calculates the coefficients of a best-fit polynomial (in a least-squares sense), such as polyfit(). For example,

can be used (using a third degree polynomial) and where

where x_lowband (k) = [0...k0-1], numTimeSlot is the number of SBR envelope time slots that exist within a frame, RATe is a constant indicating the number of QMF subband samples per slot of time (e.g. 2), ^k is a linear prediction filter coefficient (potentially obtained from the covariance method) and where

A bitstream generated in accordance with the USAC MPEG standard (sometimes referred to herein as a "USAC bitstream") includes encoded audio content and typically includes metadata indicative of each type of spectral band replication processing that apply a decoder to decode the audio content of the USAC bitstream and/or metadata that controls such spectral band replication processing and/or is indicative of at least one characteristic or parameter of at least one SBR tool and/or eSBR tool which will be used to decode the audio content of the USAC bitstream.

The term “enhanced SBR metadata” (or “eSBR metadata”) is used herein to denote metadata indicative of each type of spectral band replication processing that a decoder will apply to decode the audio content of a stream. of encoded audio bits (e.g., a USAC data stream) and/or that controls such spectral band replication processing, and/or is indicative of at least one characteristic or parameter of at least one SBR tool and/or tool eSBR that will be used to decode such audio content, but is not described or mentioned in the MPEG-4 AAC standard. An example of eSBR metadata is metadata (indicative of, or to control, spectral band replication processing) that is described or mentioned in the USAC MPEG standard but not in the MPEG-4 AAC standard. Therefore, eSBR metadata herein denotes metadata that is not SBR metadata, and SBR metadata herein denotes metadata that is not eSBR metadata.

A USAC bitstream can include both SBR metadata and eSBR metadata. More specifically, a USAC bitstream may include eSBR metadata that controls the performance of eSBR processing by a decoder, and SBR metadata that controls the performance of SBR processing by the decoder. According to typical embodiments of the present invention, the eSBR metadata (e.g., eSBR-specific configuration data) is included (in accordance with the present invention) in an MPEG-4 AAC bitstream (e.g., in the sbr_extension() container at the end of an SBR payload).

The performance of eSBR processing, during decoding of a bitstream encoded using an eSBR tool set (comprising at least one eSBR tool), by a decoder regenerates the high frequency band of the audio signal, based on replication of harmonic sequences that were truncated during encoding. Such eSBR processing typically adjusts the spectral envelope of the generated high-frequency band and applies inverse filtering, adding noise and sinusoidal components to recreate the spectral characteristics of the original audio signal.

According to typical embodiments of the invention, eSBR metadata is included (for example, a small number of control bits that are eSBR metadata are included) in one or more of the metadata segments of an encoded audio bitstream ( for example, an MPEG-4 AAC bitstream) that also includes audio data encoded in other segments (audio data segments). Typically, at least one of those metadata segments of each block of the bitstream is (or includes) a padding element (including an identifier indicating the start of the padding element), and the eSBR metadata is included in the padding element. padding after the identifier.

Figure 1 is a block diagram of an example audio processing chain (an audio data processing system), where one or more of the elements of the system may be configured in accordance with an embodiment of the present invention. The system includes the following elements, coupled as shown: encoder 1, delivery subsystem 2, decoder 3, and post-processing unit 4. In variations of the system shown, one or more of the elements are omitted, or additional audio data processing units are included.

In some implementations, the encoder 1 (which optionally includes a preprocessing unit) is configured to accept PCM (time domain) samples comprising audio content as input, and to output an encoded audio bitstream (having a format compatible with the MPEG-4 AAC standard) which is indicative of the audio content. Bitstream data that is indicative of 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 present invention, the audio bitstream output of the encoder includes eSBR metadata (and typically other metadata as well) as well as audio data.

One or more encoded audio bitstreams output from encoder 1 may be acknowledged to encoded audio delivery subsystem 2. Subsystem 2 is configured to store and/or deliver each encoded bitstream output from encoder 1. An encoded audio bitstream output from encoder 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.

Decoder 3 is configured to decode an encoded MPEG-4 AAC audio bitstream (generated by encoder 1) that it receives through subsystem 2. In some embodiments, decoder 3 is configured to extract eSBR metadata from each block of the bitstream and decode the bitstream (including performing eSBR processing using the extracted eSBR metadata) to generate decoded audio data (e.g., streams of decoded PCM audio samples). In some embodiments, the decoder 3 is configured to extract SBR metadata from the bitstream (but to ignore eSBR metadata included in the bitstream) and to decode the bitstream (including performing SBR processing using the extracted SBR metadata). to generate decoded audio data (for example, streams of decoded PCM audio samples). Typically, decoder 3 includes a buffer that stores (e.g., non-transitory) segments of the encoded audio bitstream received from subsystem 2.

The post-processing unit 4 of Figure 1 is configured to accept a stream of decoded audio data from the decoder 3 (for example, decoded PCM audio samples) and to perform post-processing thereon. The post-processing unit may also be configured to play the post-processed audio content (or the decoded audio received from the decoder 3) for playback by one or more speakers.

Figure 2 is a block diagram of an encoder 100 which is an embodiment of the audio processing unit of the invention. Any of the components or elements of the encoder 100 may be implemented as one or more processes and/or one or more circuits (e.g., ASIC, FPGA, or other integrated circuits), in hardware, software, or a combination of hardware and software. The encoder 100 includes the encoder 105, the filler/formatter stage 107, the metadata generation stage 106, and the buffer memory 109, connected as shown. Also typically, encoder 100 includes other processing elements (not shown). The encoder 100 is configured to convert an input audio bitstream to an MPEG-4 AAC encoded output bitstream.

The metadata generator 106 is coupled and configured to generate (and/or pass to step 107) metadata (including eSBR metadata and SBR metadata) to be included in step 107 in the encoded bitstream to be output from the encoder 100. .

The encoder 105 is coupled and configured to encode (e.g., performing compression thereon) the input audio data and to commit the resulting encoded audio at step 107 for inclusion in the encoded bitstream to be output from the stage 107.

Step 107 is configured to multiplex the encoded audio from encoder 105 and metadata (including eSBR metadata and SBR metadata) from generator 106 to generate the encoded bit stream to be output from step 107, preferably so that the encoded bit stream will be output from step 107. encoded bits have the format specified by one of the embodiments of the present invention.

The buffer memory 109 is configured to store (e.g., non-transitory) at least one block of the output of the encoded audio bitstream of step 107, and a sequence of the blocks of the encoded audio bitstream. is committed from buffer memory 109 as output from encoder 100 to a delivery system.

Figure 3 is a block diagram of a system that includes a decoder 200 that is an embodiment of the audio processing unit of the invention and, optionally, also a postprocessor 300 coupled thereto. Any of the components or elements of the decoder 200 and postprocessor 300 may be implemented as one or more processes and/or one or more circuits (e.g., ASIC, FPGA, or other circuits). integrated), in hardware, software or a combination of hardware and software. The decoder 200 comprises buffer memory 201, bitstream payload deformatter (parser) 205, audio decoding subsystem 202 (sometimes referred to as the "core" decoding stage or "core" decoding subsystem), eSBR processing and control bit generation step 204, connected as shown. Also typically, decoder 200 includes other processing elements (not shown).

The buffer memory (buffer) 201 stores (e.g., non-transitory) at least one block of an encoded MPEG-4 AAC audio bitstream received by the decoder 200. In operation of the decoder 200, a sequence of the blocks of the bitstream from buffer 201 to deformatter 205.

In variations of the embodiment of Figure 3 (or the embodiment of Figure 4 to be described), an APU that 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., non-transitory) at least one block of an encoded audio bitstream (e.g., an MPEG-4 AAC audio bitstream) of the same received type by the buffer 201 of Figure 3 or Figure 4 (i.e., an encoded audio bitstream that includes eSBR metadata).

Referring again to Figure 3, the deformatter 205 is coupled and configured to demultiplex each block of the bitstream to extract SBR metadata (including quantized envelope data) and eSBR metadata (and typically also other metadata) therefrom, to confirm at least the eSBR metadata and the SBR metadata to the eSBR processing step 203, and typically also to confirm other extracted metadata to the decoding subsystem 202 (and optionally also to the control bit generator 204). The deformatter 205 is also coupled and configured to extract audio data from each block of the bitstream and to commit the extracted audio data to the decoding subsystem 202 (decoding step).

The system of Figure 3 optionally also includes the postprocessor 300. The postprocessor 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., non-transitory) at least one block (or frame) of the decoded audio data received by the postprocessor 300 from the decoder 200. The processing elements of the postprocessor 300 are coupled and configured to receive and process adaptively a sequence of the blocks (or frames) of the decoded audio output of the buffer 301, using the metadata output of the decoding subsystem 202 (and/or deformatter 205) and/or control bits output from step 204 of the decoder 200.

The audio decoding subsystem 202 of the decoder 200 is configured to decode the audio data extracted by the analyzer 205 (such decoding may be referred to as a "core" decoding operation) to generate decoded audio data and to confirm the decoded audio data in eSBR processing step 203. Decoding is performed in the frequency domain and typically includes inverse quantization followed by spectral processing. Typically, a final processing step in subsystem 202 applies a frequency domain to time domain transformation to the decoded frequency domain audio data, so that the output of the subsystem is decoded time domain audio data. Step 203 is configured to apply SBR tools and eSBR tools indicated by the eSBR metadata and the eSBR (extracted by the analyzer 205) to the decoded audio data (i.e., to perform SBR and eSBR processing on the output of the subsystem 202). decoding process using the SBR and eSBR metadata) to generate the fully decoded audio data that is sent (e.g., to the post processor 300) from the decoder 200. Typically, the decoder 200 includes a memory (accessible by the subsystem 202 and the step 203) which stores the unformatted audio data and metadata output from the deformatter 205, and step 203 is configured to access the audio data and metadata (including SBR metadata and eSBR metadata) as necessary during SBR processing and eSBR. The SBR processing and eSBR processing in step 203 can be considered to be post-processing at the output of the central decoding subsystem 202. Optionally, decoder 200 also includes a final upmix subsystem (which may apply parametric stereo ("PS") tools defined in the MPEG-4 AAC standard, using PS metadata extracted by deformatter 205 and/or control bits generated in subsystem 204) which is coupled and configured to perform an upmix at the output of step 203 to generate fully decoded upmix audio that is output from the decoder 200. Alternatively, the postprocessor 300 is configured to perform an upmix at the output of decoder 200 (e.g., using PS metadata extracted by deformatter 205 and/or control bits generated in subsystem 204).

In response to the metadata extracted by the deformatter 205, the control bit generator 204 may generate control data, and the control data may be used within the decoder 200 (e.g., in a final upmix subsystem) and/or confirmed as output of decoder 200 (e.g., to postprocessor 300 for use in post-processing). In response to metadata extracted from the input bitstream (and optionally also in response to control data), step 204 may generate (and confirm to postprocessor 300) control bits indicating that the decoded audio data output of eSBR processing step 203 must undergo a specific type of post-processing. In some implementations, decoder 200 is configured to commit metadata extracted by deformatter 205 from the bitstream. input to the postprocessor 300, and the postprocessor 300 is configured to perform postprocessing on the output of decoded audio data from the decoder 200 using the metadata.

Figure 4 is a block diagram of an audio processing unit ("APU") 210 which is another embodiment of the APU audio processing unit 210 of the invention is a legacy decoder that is not configured to perform eSBR processing. Any of the components or elements of APU 210 may be implemented as one or more processes and/or one or more circuits (e.g., ASIC, FPGA, or other integrated circuits), in hardware, software, or a combination of hardware and software. The APU 210 comprises buffer memory 201, bitstream payload deformatter (parser) 215, audio decoding subsystem 202 (sometimes referred to as the "core" decoding stage or "core" decoding subsystem), and SBR processing, connected as shown. Also typically, APU 210 includes other processing elements (not shown). The APU 210 may represent, for example, an audio encoder, decoder, or transcoder.

The elements 201 and 202 of the APU 210 are identical to the identically numbered elements of the decoder 200 (of Figure 3) and the previous description of them will not be repeated. In operation of the APU 210, a sequence of blocks of an encoded audio bitstream (an MPEG-4 AAC bitstream) received by the APU 210 is committed from the buffer 201 to the deformatter 215.

The deformatter 215 is coupled and configured to demultiplex each block of the bitstream to extract SBR metadata (including quantized envelope data) and typically also other metadata thereof, but to ignore eSBR metadata that may be included in the bitstream. bits according to any embodiment of the present invention. The deformatter 215 is configured to commit at least the SBR metadata in the SBR processing step 213. The deformatter 215 is also coupled and configured to extract audio data from each block of the bitstream and to commit the extracted audio data to the decoding subsystem 202 (decoding step).

The audio decoding subsystem 202 of the decoder 200 is configured to decode the audio data extracted by the deformatter 215 (such decoding may be referred to as a "core" decoding operation) to generate decoded audio data and to confirm the decoded audio data in SBR processing step 213. Decoding is performed in the frequency domain. Typically, a final processing step in subsystem 202 applies a frequency domain to time domain transformation to the decoded frequency domain audio data, so that the output of the subsystem is decoded time domain audio data. Step 213 is configured to apply SBR tools (but not eSBR tools) indicated by the SBR metadata (extracted by the deformatter 215) to the decoded audio data (i.e., to perform SBR processing on the output of the decoding subsystem 202). using the SBR metadata) to generate the fully decoded audio data that is sent (e.g., to the post processor 300) from the APU 210. Typically, the APU 210 includes a memory (accessible by subsystem 202 and stage 213) that stores the deformatted audio data and metadata output from the deformatter 215, and step 213 is configured to access the audio data and metadata (including SBR metadata) as needed during SBR processing. The SBR processing in step 213 can be considered to be post-processing at the output of the central decoding subsystem 202. Optionally, the APU 210 also includes a final upmix subsystem (which can apply parametric stereo ("PS") tools defined in the MPEG-4 AAC standard, using PS metadata extracted by the deformatter 215) that is coupled and configured to perform upmix at the output of step 213 to generate upmixed, fully decoded audio that is output from the APU 210. Alternatively, a postprocessor is configured to perform an upmix at the output of the APU 210 (e.g., using metadata PS extracted by the deformatter 215 and/or control bits generated in the APU 210).

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

According to some embodiments, eSBR metadata is included (e.g., a small number of control bits that are eSBR metadata are included) in an encoded audio bitstream (e.g., an MPEG-4 AAC bitstream), so that legacy decoders (that are not configured to parse the eSBR metadata, or to use any eSBR tool to which the eSBR metadata belongs) can ignore the eSBR metadata but nevertheless decode the bitstream to the extent as possible without using the eSBR metadata or any eSBR tool to which the eSBR metadata belongs, typically without any significant penalty in the quality of the decoded audio. However, eSBR decoders configured to analyze the bitstream to identify the eSBR metadata, and to use at least one eSBR tool in response to the eSBR metadata, will enjoy the benefits of using at least one such eSBR tool. Therefore, embodiments of the invention provide a means to efficiently transmit enhanced spectral band replication (eSBR) control data or metadata in a backward-compatible manner.

Typically, the eSBR metadata in the bitstream is indicative of (for example, is indicative of at least one characteristic or parameter of) one or more of the following eSBR tools (which are described in the USAC MPEG standard, and which may or not having been applied by an encoder during the generation of the bitstream): - harmonic transposition; and

- additional QMF patching preprocessing (pre-flattening).

For example, the eSBR metadata included in the bitstream may be indicative of parameter values (described in the USAC MPEG standard and in the present disclosure): sbrPatchingMode[ch], sbrOversamplingFlag[ch], sbrPitchInBins[ch], sbrPitchInBins[ch] and bs_sbr_preprocessing.

Herein, the notation X[ch], where X is some parameter, denotes that the parameter belongs to the channel ("ch") of the audio content of an encoded bitstream to be decoded. For simplicity, we sometimes omit the [ch] expression and assume that the relevant parameter belongs to an audio content channel. In this document, the notation X[ch][env], where encoded bits to be decoded. For simplicity, we sometimes omit the expressions [env] and [ch], and assume that the relevant parameter belongs to an SBR envelope of an audio content channel.

During decoding of an encoded bitstream, the harmonic transposition performance during an eSBR processing step of the decoding (for each channel, "ch", of the audio content indicated by the bitstream) is controlled by the following eSBR metadata parameters: sbrPatchingMode[ch]: sbrOversamplingFlag[ch]; sbrPitchInBinsFlag[ch]; and sbrPitchInBins[ch].

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

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

The value "sbrPitchlnBinsFlag[ch]" controls the interpretation of the sbrPitchInBins[ch] parameter: 1 indicates that the value in sbrPitchlnBins[ch] is valid and greater than zero; 0 indicates that the value of sbrPitchlnBins[ch] is set to zero. The value "sbrPitchInBins[ch]" controls the addition of cross-product terms in the SBR harmonic transponder. The sbrPitchinBins[ch] value is an integer value in the range [0.127] and represents the distance measured in frequency bins for a 1536-line DFT acting on the core encoder sampling rate.

In the case where an MPEG-4 AAC bitstream is indicative of a pair of SBR channels whose channels are not coupled (rather than a single SBR channel), the bitstream is indicative of two instances of the above syntax ( for harmonic or non-harmonic transposition), one for each channel of the sbr_channeLpair_element().

The eSBR tool's harmonic transposition typically improves the quality of decoded music signals at relatively low crossover frequencies. Nonharmonic transposition (i.e., inherited spectral patching) typically enhances speech signals. Therefore, a starting point in deciding which type of transposition is preferable for encoding specific audio content is to select the transposition method depending on the voice/music detection with harmonic transposition employed in the musical content and the spectral patching. in speed content.

The performance of pre-flattening during eSBR processing is controlled by the value of a one-bit eSBR metadata parameter known as "bs_sbr_preprocessing", in the sense that pre-flattening is performed or not depending on the value of this single bit. When using the SBR QMF patching algorithm, as described in section 4.6.18.6.3 of the MPEG-4 AAC standard, the pre-flattening step can be performed (when indicated by the "bs_sbr_preprocessing" parameter) in an effort to avoid discontinuities in the shape of the spectral envelope of a high-frequency signal that is input to a subsequent envelope adjuster (the envelope adjuster performs another stage of eSBR processing). Pre-flattening typically improves the performance of the subsequent envelope adjustment stage, resulting in a high-band signal that is perceived as more stable.

The overall bit rate requirement for including in the eSBR metadata of an MPEG-4 AAC bitstream indicative of the eSBR tools mentioned above (harmonic transposition and pre-flattening) is expected to be on the order of a few hundred bits per second because only differential control data necessary to perform eSBR processing is transmitted in accordance with some embodiments of the invention. Legacy decoders may ignore this information because it is included in a backwards-compatible manner (as will be explained later). Therefore, the detrimental effect on bitrate associated with the inclusion of eSBR metadata is negligible, for several reasons, including the following:

- the bitrate penalty (due to including eSBR metadata) is a very small fraction of the total bitrate because only the differential control data necessary to perform eSBR processing is transmitted (and not a simultaneous transmission of SBR control data); and

- the setting of SBR-related control information does not typically depend on the details of the transposition. Examples of when control data depends on transponder operation are described later in this application.

Thus, embodiments of the invention provide a means to efficiently transmit enhanced spectral band replication (eSBR) control data or metadata in a backward-compatible manner. This efficient transmission of eSBR control data reduces memory requirements in decoders, encoders and transcoders employing aspects of the invention, while having no tangible adverse effect on the bit rate. Furthermore, the complexity and processing requirements associated with performing eSBR according to embodiments of the invention are also reduced because the SBR data must be processed only once and not broadcast simultaneously, which would be the case if eSBR were processed as a completely separate object type in MPEG-4 AAC rather than being integrated into the MPEG-4 AAC codec in a backward compatible manner.

Next, with reference to Figure 7, we describe elements of a block ("raw_data_block") of an MPEG-4 AAC bitstream where eSBR metadata is included in accordance with some embodiments of the present invention. Figure 7 is a diagram of a block (a "raw_data_block") of the MPEG-4 aAc bitstream, showing some of its segments.

A block of an MPEG-4 AAC bitstream may include at least one "single_channel_element()" (for example, the single channel element shown in Figure 7), and/or at least one "channel_pair_element() " (not specifically shown in Figure 7 although it may be present), including audio data from an audio program. The block may also include a series of "filLelements" (e.g., filler element 1 and/or filler element 2 of Figure 7) that include data (e.g., metadata) related to the program. Each "single_channel_element()" includes an identifier (e.g., "ID1" of Figure 7) that indicates the start of a single channel element, and may include audio data indicative of a different channel of a multichannel audio program. Each channel pair element includes an identifier (not shown in Figure 7) that indicates the start of a channel pair element, and may include audio data indicative of two channels of the program.

A fill_element (referred to herein as a fill element) of an MPEG-4 AAC bitstream includes an identifier ("ID2" of Figure 7) indicating the start of a fill element and fill data after the identifier. The ID2 identifier may consist of a three-bit unsigned integer transmitted most significant bit first ("uimsbf") that has a value of 0x6. The padding data may include 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. There are several types of extension payloads and they are identified by the "extension_type" parameter, which is a four-bit unsigned integer transmitted most significant bit first ("uimsbf").

The padding data (e.g., an extension payload thereof) may include a header or identifier (e.g., "header" of Figure 7) that indicates a segment of padding data that is indicative of an SBR object. (i.e. the header initializes an "SBR object" type, called sbr_extension_data() in the MPEG-4 AAC standard). For example, a Spectral Band Replication (SBR) extension payload is identified by the value of '1101' or '1110' for the extension_type field in the header, with the identifier '1101' identifying an extension payload. with SBR data and '1110' which identifies an extension payload with SBR data with a cyclic redundancy check (CRC) to verify the accuracy of the SBR data.

When the header (for example, the extension_type field) initializes an SBR object type, the SBR metadata (sometimes referred to herein as "spectral band replication data" and called sbr_data() in the MPEG-4 AAC standard) follow the header, and at least one spectral band replication extension element (for example, the "SBR extension element" of filler element 1 of Figure 7) may follow the SBR metadata. Such a spectral band replication extension element (a segment of the bitstream) is called a "sbr_extension()" container in the MPEG-4 AAC standard. An extension element Spectral band replication optionally includes a header (e.g., "SBR extension header" of filler element 1 of Figure 7).

The MPEG-4 AAC standard contemplates that a spectral band replication extension element may include PS (parametric stereo) data for audio data of a program. The MPEG-4 AAC standard provides that when the header of a filler element (for example, an extension payload thereof) initializes an SBR object type (such as "header1" in Figure 7) and an element The spectral band replication extension payload of the padding element includes PS data, the padding element (for example, the extension payload thereof) includes spectral band replication data and a parameter "bs_extension_id" whose value (i.e. bs_extension_id = 2) indicates that PS data is included in a filler element spectral band replication extension element.

In accordance with some embodiments of the present invention, eSBR metadata (e.g., an indicator indicative of whether enhanced spectral band replication (eSBR) processing will be performed on the audio content of the block) is included in an extension element. of spectral band replication of a filler element. For example, such an indicator is indicated in filler element 1 of Figure 7, where the indicator appears after the header (the "SBR extension header" of filler element 1) of the "SBR extension element" of element 1 of stuffed. Optionally, such an indicator and additional eSBR metadata are included in a spectral band replication extension element after the header of the spectral band replication extension element (for example, in the SBR extension element of filler element 1 in Figure 7, after the SBR extension header). According to some embodiments of the present invention, a padding element that includes eSBR metadata also includes a parameter "bs_extension_id" whose value (e.g., bs_extension_id = 3) indicates that the eSBR metadata is included in the padding element and that the eSBR processing must be performed on the audio content of the relevant block.

According to some embodiments of the invention, the eSBR metadata is included in a padding element (for example, padding element 2 of Figure 7) of an MPEG-4 AAC bitstream that is not a bit extension element. spectral band replication (SBR extension element) of the filler element. This is because filler elements that contain an extension_payload() with SBR data or SBR data with a CRC do not contain any other extension payloads of any other extension type. Therefore, in embodiments where the eSBR metadata stores its own extension payload, a separate padding element is used to store the eSBR metadata. Such a padding element includes an identifier (e.g., "ID2" of Figure 7) indicating the start of a padding element and padding data after the identifier. The padding data may include 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., an extension payload thereof) includes a header (e.g., "header2" of the padding element 2 of Figure 7) that is indicative of an eSBR object (i.e., the header initializes an enhanced spectral band replication (eSBR) object type), and the padding data (e.g., an extension payload thereof) includes eSBR metadata after the header. For example, padding element 2 of Figure 7 includes such a header ("header2") and also includes, after the header, eSBR metadata (i.e., the "flag" in padding element 2, which indicates whether replication (eSBR) should be performed on the audio content of the block). Optionally, additional eSBR metadata is also included in the padding data of padding element 2 of Figure 7, after header2. In the embodiments described in this paragraph, the header (for example, header2 of Figure 7) has an identification value that is not one of the conventional values specified in Table 4.57 of the MPEG-4 AAC standard, and in change is indicative of an eSBR extension payload (so that the extension_type field in the header indicates that the padding data includes eSBR metadata).

In a first class of embodiments, the invention is an audio processing unit (for example, a decoder), comprising:

a memory (e.g., buffer 201 of Figure 3 or 4) 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 memory and configured to demultiplex at least a portion of said 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), coupled and configured to decode at least a portion of the audio content of said block of the bitstream, where The block includes:

a padding element, which includes an identifier indicating a start of the padding element (for example, the identifier "id_syn_ele" having the value 0x6, from Table 4.85 of the MPEG-4 AAC standard), and padding data after the identifier, where the padding data includes:

at least one flag that identifies whether enhanced spectral band replication (eSBR) processing should be performed on the audio content of the block (for example, using spectral band replication data and eSBR metadata included in the block).

The flag is the eSBR metadata and an example of the flag is the sbrPatchingMode flag. Another example of the indicator is the harmonicSBR indicator. Both flags indicate whether a baseline form of spectral band replication or an enhanced form of spectral replication is to be performed on the audio data in the block. The basic form of spectral replication is spectral patching, and the enhanced form of spectral band replication is harmonic transposition.

In some embodiments, the fill data also includes additional eSBR metadata (i.e., eSBR metadata other than the indicator).

The memory may be a buffer (e.g., an implementation of the buffer memory 201 of Figure 4) that stores (e.g., non-transitory) at least said block of the encoded audio bitstream.

It is estimated that the performance complexity of eSBR processing (using eSBR harmonic transposition and pre-flattening) by an eSBR decoder during decoding of an MPEG-4 AAC bitstream that includes eSBR metadata (indicative of these eSBR tools) would be following (for typical decoding with the indicated parameters):

• Harmonic transposition (16 kbps, 14400/28800 Hz)

° Based on DFT: 3.68 WMOPS (weighted million operations per second);

o Based on QMF: 0.98 WMOPS;

• QMF patching preprocessing (pre-flattening): 0.1WMOPS.

DFT-based transpose is known to typically perform better than QMF-based transpose for transients.

According to some embodiments of the present invention, a padding element (of an encoded audio bitstream) that includes eSBR metadata also includes a parameter (e.g., a "bs_extension_id" parameter) whose value (e.g., bs_extension_id = 3) indicates that the eSBR metadata is included in the padding element and that the eSBR processing must be performed on the audio content of the relevant block, and/or a parameter (for example, the same parameter "bs_extension_id") whose value ( for example, bs_extension_id = 2) indicates that a filler element's sbr_extension() container includes pS data. For example, as indicated in Table 1 below, such a parameter having the value bs_extension_id = 2 may indicate that a fill element sbr_extension() container includes PS data, and such a parameter having the value bs_extension_id = 3 may indicate that a filler element's sbr_extension() container includes eSBR metadata:

Table 1

In accordance with some embodiments of the invention, the syntax of each spectral band replication extension element that includes eSBR metadata and/or PS data is as set forth in Table 2 below (where "sbr_extension()" denotes a container which is the spectral band replication extension element, "bs_extension_id" is as described in table 1 above, "ps_data" denotes PS data and "esbr_data" denotes eSBR metadata):

Table 2

In an example embodiment, esbr_data() referenced in Table 2 above is indicative of the values of the following metadata parameters:

1. The one-bit metadata parameter, "bs_sbr_preprocessing"; and

2. For each channel ("ch") of audio content of the encoded bitstream to be decoded, each of the parameters described above: "sbrPatchingMode[ch]"; "sbrOversamplingFlag[ch]"; "sbrPitchlnBinsFlag[ch]"; and "sbrPitchlnBins[ch]".

For example, in some embodiments, esbr_data() may have the syntax indicated in Table 3, to indicate these metadata parameters:

Table 3

The above syntax allows for an efficient implementation of an improved form of spectral band replication, such as harmonic transposition, as an extension of a legacy decoder. Specifically, the eSBR data in Table 3 includes only those parameters necessary to perform the enhanced form of spectral band replication that are not already supported in the bitstream or directly derivable from parameters already supported in the bitstream. All other parameters and processing data necessary to perform the enhanced form of spectral band replication are extracted from pre-existing parameters at locations already defined in the bitstream.

For example, an MPEG-4 HE-AAC or HE-AAC v2 compliant decoder can be expanded to include an improved form of spectral band replication, such as harmonic transposition. This enhanced form of spectral band replication is in addition to the base form of spectral band replication that is already supported by the decoder. In the context of an MPEG-4 HE-AAC or HE-AAC v2 compliant decoder, this base form of spectral band replication is the QMF spectral patching SBR tool as defined in section 4.6.18 of the MPEG-4 standard. 4 AAC.

By performing the enhanced form of spectral band replication, an extended HE-AAC decoder can reuse many of the bitstream parameters already included in the bitstream SBR extension payload. Specific parameters that can be reused include, for example, the various parameters that determine the master frequency band table. These parameters include bs_start_freq (parameter that determines the start of the master frequency table), bs_stop_freq (parameter that determines the stop of the master frequency table), bs_freq_scale (parameter that determines the number of frequency bands per octave), and bs_alter_scale ( parameter that modifies the scale of the frequency bands). Parameters that can be reused also include parameters that determine the noise band table (bs_noise_bands) and limiter band table parameters (bs_limiter_bands). Accordingly, in various embodiments, at least some of the equivalent parameters specified in the USAC standard are omitted from the bitstream, thereby reducing the control overhead in 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, for example, the scale factor of the EOrigMapped envelope. However, the equivalent parameter specified in the USAC standard typically has a different value, which is "tuned" for enhanced SBR processing defined in the USAC standard rather than SBR processing defined in the AAC standard.

To improve the subjective quality of audio content with harmonic frequency structure and strong tonal characteristics, particularly at low bit rates, enabling enhanced SBR is recommended. The values of the corresponding bitstream element (i.e., esbr_data()), which controls these tools, can be determined in the encoder by applying a signal-dependent sorting mechanism. Generally, use of the harmonic patching method (sbrPatchingMode == 1) is preferable for encoding music signals at very low bit rates, where the core codec may be considerably limited in audio bandwidth. This is especially true if these signals include a pronounced harmonic structure. In contrast, the use of the normal SBR patching method is preferred for speech and mixed signals, as it provides better preservation of the temporal structure in the speech.

To improve the performance of the harmonic transponder, a preprocessing step (bs_sbr_preprocessing == 1) can be activated that strives to avoid introducing spectral discontinuities of the signal entering the subsequent envelope adjuster. The tool's operation is beneficial for signal types where the coarse spectral envelope of the low-band signal used for high-frequency reconstruction shows large level variations.

To improve the transient response of SBR harmonic patching, signal adaptive frequency domain oversampling (sbrOversamplingFlag == 1) can be applied. Since signal adaptive frequency-domain oversampling increases the computational complexity of the transponder, but only provides benefits for frames containing transients, the use of this tool is controlled by the bitstream element, which is transmitted once per frame and by independent SBR channel.

A decoder operating in the proposed enhanced SBR mode typically needs to be able to switch between legacy and enhanced SBR patching. Therefore, a delay can be introduced that can be as long as the duration of a core audio frame, depending on the decoder configuration. Typically, the delay for both legacy and enhanced SBR patching will be similar.

In addition to the numerous parameters, an extended HE-AAC decoder may also reuse other data elements when performing an enhanced form of spectral band replication in accordance with embodiments of the invention. For example, envelope data and background noise data can also be extracted from the bs_data_env (envelope scale factors) and bs_noise_env (background noise scale factors) data and used during the enhanced form of replication. spectral band.

In essence, these embodiments exploit the configuration parameters and envelope data already supported by a legacy HE-AAC or HE-AAC v2 decoder in the SBR extension payload to enable an enhanced form of spectral band replication that requires the least possible amount of additional transmitted data. The metadata was originally tuned for a base form of HFR (e.g., the spectral translation operation of SBR), but according to embodiments, it is used for an enhanced form of HFR (e.g., the harmonic transposition of eSBR). . As explained above, metadata generally represents operational parameters (e.g., envelope scaling factors, background noise scaling factors, time/frequency grid parameters, sinusoid addition information, variable crossover over frequency/band, inverse filtering mode, envelope resolution, smoothing mode, frequency interpolation mode) tuned and intended for use with the base form of HFR (e.g. linear spectral translation). However, this metadata, combined with additional metadata parameters specific to the enhanced form of HFR (e.g., harmonic transposition), can be used to efficiently and effectively process audio data using the enhanced form of HFR.

Consequently, extended decoders that support an improved form of spectral band replication can be created in a very efficient manner by relying on already defined bitstream elements (e.g., those in the SBR extension payload) and adding only the parameters required to support the enhanced form of spectral band replication (in a filler extension payload). This data reduction feature combined with placing the newly added parameters in a reserved data field, such as an extension container, substantially reduces the barriers to creating a decoder that supports an improved form of spectral band replication by ensuring that the bitstream is backward compatible with the legacy decoder that does not support the enhanced form of spectral band replication. It will be appreciated that the reserved data field is a backward compatible data field, that is, it is a data field that is already supported by previous decoders, such as legacy HE-AAC or HE-AAC v2 decoders. Similarly, the extension container is backward compatible, that is, it is an extension container that is already supported by previous decoders, such as legacy HE-AAC or HE-AAC v2 decoders.

In Table 3, the number in the right column indicates the number of bits of the corresponding parameter in the left column.

In some embodiments, the SBR object type defined in MPEG-4 AAC is updated to contain the SBR tool and aspects of the enhanced SBR (eSBR) tool as indicated in the SBR extension element (bs_extension_id == EXTENSION_ID_ESBR). If a decoder detects this SBR extension element, the decoder employs the signaled aspects of the enhanced SBR tool.

In some embodiments, the invention is a method that includes an audio data encoding step to generate an encoded bitstream (e.g., an MPEG-4 AAC bitstream), including eSBR metadata in at least one segment of at least one block of the encoded bitstream and audio data in at least one other segment of the block. In typical embodiments, the method includes a step of multiplexing the audio data with the eSBR metadata in each block of the encoded bitstream. In typical decoding of the encoded bitstream in an eSBR decoder, the decoder extracts the eSBR metadata from the bitstream (including parsing and demultiplexing the eSBR metadata and audio data) and uses the eSBR metadata to process the data. to generate a stream of decoded audio data.

Another aspect of the invention is an eSBR decoder configured to perform eSBR processing (e.g., using at least one of the eSBR tools known as harmonic transposition or pre-flattening) during decoding of an encoded audio bitstream (e.g. an MPEG-4 AAC bitstream) that does not include eSBR metadata. An example of such a decoder will be described with reference to Figure 5.

The eSBR decoder 400 of Figure 5 includes buffer memory 201 (which is identical to memory 201 of Figures 3 and 4), bitstream payload deformatter 215 (which is identical to deformatter 215 of Figure 4), the audio decoding subsystem 202 (sometimes called the "core" decoding stage or the "core" decoding subsystem, and which is identical to the decoding subsystem 202 of Figure 3), the data generation subsystem 401 eSBR control and eSBR processing stage 203 (which is identical to stage 203 of Figure 3), connected as shown. Also typically, decoder 400 includes other processing elements (not shown).

In operation of the decoder 400, a sequence of blocks of an encoded audio bitstream (an MPEG-4 AAC bitstream) received by the decoder 400 is committed from the buffer 201 to the deformatter 215. The deformatter 215 is coupled and configured to demultiplex each block of the bitstream to extract SBR metadata (including quantized envelope data) and typically also other metadata therefrom. The deformatter 215 is configured to commit at least the SBR metadata in the eSBR processing step 203. The deformatter 215 is also coupled and configured to extract audio data from each block of the bitstream and to commit the extracted audio data to the decoding subsystem 202 (decoding step).

The audio decoding subsystem 202 of the decoder 400 is configured to decode the audio data extracted by the deformatter 215 (such decoding may be referred to as a "core" decoding operation) to generate decoded audio data and to confirm the decoded audio data in eSBR processing step 203. Decoding is performed in the frequency domain. Typically, a final processing step in subsystem 202 applies a frequency domain to time domain transformation to the data. decoded frequency domain audio data, so that the output of the subsystem is decoded time domain audio data. Step 203 is configured to apply SBR tools (and eSBR tools) indicated by the SBR metadata (extracted by the deformatter 215) and by the eSBR metadata generated in subsystem 401, to the decoded audio data (i.e., to perform the SBR and eSBR processing at the output of the decoding subsystem 202 using the SBR and eSBR metadata) to generate the fully decoded audio data that is output from the decoder 400. Typically, the decoder 400 includes a memory (accessible by the subsystem 202 and step 203) which stores the deformatted audio data and metadata output from deformatter 215 (and optionally also subsystem 401), and step 203 is configured to access the audio data and metadata as needed during SBR processing and eSBR. The SBR processing in step 203 can be considered to be post-processing at the output of the central decoding subsystem 202. Optionally, decoder 400 also includes a final upmix subsystem (which may apply parametric stereo ("PS") tools defined in the MPEG-4 AAC standard, using PS metadata extracted by deformatter 215) that is coupled and configured to perform upmix at the output of stage 203 to generate fully decoded upmix audio that is output from the APU 210.

Parametric stereo is an encoding tool that represents a stereo signal using a linear downmix of the left and right channels of the stereo signal and sets of spatial parameters that describe the stereo image. Parametric stereo typically employs three types of spatial parameters: (1) interchannel intensity differences (IID) that describe the intensity differences between channels; (2) interchannel phase differences (IPD) that describe the phase differences between channels; and (3) inter-channel coherence (ICC) which describes the coherence (or similarity) between channels. Coherence can be measured as the maximum of the cross-correlation as a function of time or phase. These three parameters generally allow a high-quality reconstruction of the stereo image. However, the IPD parameters only specify the relative phase differences between the channels of the stereo input signal and do not indicate the distribution of these phase differences in the left and right channels. Therefore, a fourth type of parameter that describes an overall phase shift or an overall phase difference (OPD) can be additionally used. In the stereo reconstruction process, consecutive window segments of both the received downmix signal, s[n], and a decorrelated version of the received downmix, d[n], are processed along with spatial parameters to generate the left (lk(n)) and right (rk(n)) reconstructed the signals according to:

where H ¹¹ , H ¹² , H ²¹ and H ²² are defined by the stereo parameters. The signals lk(n) and rk(n) are finally transformed back to the time domain by a frequency-to-time transformation.

The control data generation subsystem 401 of 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 metadata eSBR of any type included in encoded audio bitstreams according to other embodiments of the invention) in response to at least one result of the detection step. The eSBR control data is confirmed at step 203 to trigger the application of individual eSBR tools or combinations of eSBR tools upon detecting a specific property (or combination of properties) of the bitstream and/or to control the application of such eSBR tools. . For example, to control the performance of eSBR processing using harmonic transposition, some embodiments of the control data generation subsystem 401 would include: a music detector (e.g., a simplified version of a conventional music detector) to configure the parameter sbrPatchingMode[ch] (and confirm the parameter set in step 203) in response to detecting that the bitstream is or is not indicative of music; a transient detector for setting the sbrOversamplingFlag[ch] parameter (and confirming the parameter set in step 203) in response to detecting the presence or absence of transients in the audio content indicated by the bitstream; and/or a pitch detector to set the parameters sbrPitchInBinsFlag[ch] and sbrPitchlnBins[ch] (and confirm the parameters set in step 203) in response to detecting the pitch of the audio content indicated by the bitstream. Other aspects of the invention are the audio bitstream decoding methods performed by any embodiment of the decoder of the invention described in this paragraph and the previous paragraph.

Aspects of the invention include an encoding or decoding method of the type for which any embodiment of the APU, system or device of the invention is configured (e.g., programmed). Other aspects of the invention include a system or device configured (e.g., programmed) to perform any embodiment of the method of the invention, and a computer-readable medium (e.g., a disk) that stores code (e.g., in a non-transparent manner). transient) to implement any embodiment of the method of the invention or steps thereof. For example, the system of the invention may be or include a purpose processor. general programmable, digital signal processor or microprocessor, programmed with software or firmware and/or otherwise configured to perform any of a variety of operations on data, including an embodiment of the method of the invention or steps thereof. Such a general purpose processor may be or include a computer system that includes an input device, memory, and processing circuitry programmed (and/or otherwise configured) to perform an embodiment of the method of the invention (or steps thereof). ) in response to the data confirmed therein.

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 specified, any algorithms or processes included as part of the invention are not intrinsically related to any particular computer or other device. In particular, various general-purpose machines may be used with programs written in accordance with the teachings herein, or it may be more convenient to construct more specialized apparatus (e.g., integrated circuits) to perform the required steps of the method. Therefore, the invention can be implemented in one or more computer programs that run on one or more programmable computer systems (for example, an implementation of any of the elements of Figure 1, or the encoder 100 of Figure 2 (or a element thereof), or decoder 200 of Figure 3 (or an element thereof), or decoder 210 of Figure 4 (or an element thereof), or decoder 400 of Figure 5 (or an element thereof)) each of which comprises at least one processor, at least one data storage system (including memory and/or volatile and non-volatile storage elements), at least one input device or port and at least one device or port of departure. Program code is applied to input data to perform the functions described herein and generate output information. The output information is applied to one or more output devices, in a known manner.

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

For example, when implemented by computer software instruction sequences, various functions and steps of embodiments of the invention may be implemented by multithreaded software instruction sequences executing on suitable digital signal processing hardware, in which case the various devices , steps and functions of the embodiments may correspond to portions of the software instructions.

Each of these computer programs is preferably stored or downloaded on a storage medium or device (for example, memory or solid state medium, or magnetic or optical medium) readable by a general or special purpose programmable computer, to configure and operate the computer when the storage medium or device is read by the computer system to perform the procedures described herein. The system of the invention may also be implemented as a computer-readable storage medium, configured with (i.e., storing) a computer program, where the storage medium so configured causes a computer system to operate in a specific and predefined manner to perform the functions described in this document.

Various embodiments of the invention have been described. However, it will be understood that various modifications can be made. Numerous modifications and variations of the present invention are possible in light of the above teachings. For example, to facilitate efficient implementations, phase shifts can be used in combination with complex QMF analysis and synthesis filter banks. The analysis filter bank is responsible for filtering the low band time domain signal generated by the central decoder into a plurality of subbands (e.g., QMF subbands). The synthesis filter bank is responsible for combining the regenerated high band produced by the selected HFR technique (as indicated by the received sbrPatchingMode parameter) with the decoded low band to produce a wideband output audio signal. However, a given filter bank implementation operating in a certain sample rate mode, for example, normal dual-rate operation or downsampled SBR mode, should not have phase shifts that depend on the bitstream. The QMF banks used in SBR are a complex exponential extension of the cosine modulated filter bank theory. It can be shown that alias cancellation constraints become obsolete when the cosine modulation filter bank is expanded with complex exponential modulation. Therefore, for SBR QMF banks, both the analysis filters, hk(n) and the synthesis filters, fk(n), can be defined by:

where p ⁰ (n) is a real-valued symmetric or asymmetric prototype filter (typically a low-pass prototype filter), M denotes the number of channels, and N is the order of the prototype filter. The number of channels used in the analysis filter bank may be different than the number of channels used 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 operates the synthesis filter bank in downsampling mode, the synthesis filter bank can have only 32 channels. Since the subband samples of the filter bank are complex in value, a possibly channel-dependent additive phase shift step can be added to the analysis filter bank. These additional phase shifts must be compensated for before the synthesis filter bank. While phase change terms can in principle be of arbitrary values without destroying the operation of the QMF analysis/synthesis chain, they can also be limited to certain values for conformance checking. The SBR signal will be affected by the choice of phase factors, while the low pass signal coming from the central decoder will not. The audio quality of the output signal is not affected.

The prototype filter coefficients, p ⁰ (n), can be defined with a length, L, of 640, as shown in Table 4 below.

Table 4

The prototype filter, po(n), can also be derived from Table 4 using one or more mathematical operations such as rounding, subsampling, interpolation, and decimation.

Although the setting of SBR-related control information does not typically depend on the details of the transposition (as explained above), in some embodiments, certain elements of the control data may be concurred in the eSBR extension container (bs_extension_id = = EXTENSION_ID_ESBR) to improve the quality of the regenerated signal. Some of the simultaneous elements may include background noise data (for example, background noise scaling factors and a parameter indicating the direction, either in the frequency direction or in the time direction, of the delta coding for each background noise), the reverse filtering data (for example, a parameter indicating the reverse filtering mode selected from no reverse filtering, a low level of reverse filtering, an intermediate level of reverse filtering, and a high level of reverse filtering ) and missing harmonic data (for example, a parameter indicating whether a sinusoid should be added to a specific frequency band of the regenerated high band). All of these elements are based on a synthesized emulation of the decoder transponder performed in the encoder and therefore, if tuned correctly for the selected transponder, can increase the quality of the regenerated signal.

Specifically, in some embodiments, the missing harmonics and inverse filtering control data are transmitted in the eSBR extension container (along with the other bitstream parameters in Table 3) and tuned to the eSBR harmonic transponder. eSBR. The additional bit rate required to transmit these two classes of metadata for the eSBR harmonic transponder is relatively low. Therefore, sending reverse filtering control data and/or tuned missing harmonics into the eSBR extension container will increase the quality of the audio produced by the transponder and only minimally affect the bit rate. To ensure backward compatibility with legacy decoders, tuned parameters for SBR spectral translation operation can also be sent in the bitstream as part of the SBR control data using implicit or explicit signaling.

It should be understood that within the scope of the appended claims, the invention may be practiced in various ways. The reference numbers contained in the following claims are for illustrative purposes only and should not be used to interpret or limit the claims in any way.

Claims (7)

1. - A method for performing high frequency reconstruction of an audio signal, the method comprising: receiving an encoded audio bitstream, the encoded audio bitstream including audio data representing a low band portion of the audio signal and high-frequency reconstruction metadata; decoding the audio data to generate a decoded low band audio signal; extract from the encoded audio bitstream the high frequency reconstruction metadata, the high frequency reconstruction metadata including the operating parameters for a high frequency reconstruction process, the operating parameters including a patch mode parameter located in a container backward compatible extension of the encoded audio bitstream, where a first value of the patching mode parameter indicates spectral translation and a second value of the patching mode parameter indicates harmonic transposition by phase vocoder frequency expansion; filtering the decoded low band audio signal to generate a filtered low band audio signal; regenerate a high-band portion of the audio signal using the filtered low-band audio signal and the high-frequency reconstruction metadata, where the regeneration includes the spectral translation if the patch mode parameter is the first value and the regeneration includes harmonic transposition by frequency expansion of the phase vocoder if the patch mode parameter is the second value; and combining the filtered low band audio signal with the regenerated high band portion to form a wide band audio signal; wherein the encoded audio bitstream further includes a padding element with an identifier indicating a start of the padding element and padding data after the identifier, wherein the padding data includes the backward compatible extension container, wherein the padding data include an extension payload, the extension payload includes spectral band replication extension data, and the extension payload is identified by a four-bit unsigned integer transmitted most significant bit first and having a value from '1101' or '1110'; wherein the high frequency reconstruction metadata includes envelope scale factors. 2. - The method of claim 1, wherein the filtering is performed using a bank of analysis filters that includes analysis filters, hk (n), which are modulated versions of a prototype filter, p ⁰ (n), according to with:

where p ⁰ (n) is a real-valued symmetric or asymmetric prototype filter, M is a number of channels in the analysis filter bank, and N is an order of the prototype filter. 3.- The method of claim 2, where the prototype filter, p ⁰ (n), is derived from the coefficients in the following table.

4. - The method of claim 2, wherein the prototype filter, po(n), is derived from the coefficients of the table defined in claim 3 by means of one or more mathematical operations selected from the group consisting of rounding, subsampling, interpolation or decimation. 5. - The method of any one of claims 1 to 4, wherein a phase shift is added to the filtered low-band audio signal after filtering and compensated before combining to reduce the complexity of the method. 6. - A non-transitory computer-readable medium that contains instructions that when executed by a processor perform the method of any of claims 1 to 5. 7. - 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-band portion of the audio signal and high-frequency reconstruction metadata; a central audio decoder (202) for decoding the audio data to generate a decoded low band audio signal; a deformatter (205, 215) for extracting from the encoded audio bitstream the high frequency reconstruction metadata, the high frequency reconstruction metadata including the operating parameters for a high frequency reconstruction process, the operating parameters including a parameter patch mode located in a backward compatible extension container of the encoded audio bitstream, where 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 expansion of phase vocoder; a bank (203, 213) of analysis filters for filtering the decoded low band audio signal to generate a filtered low band audio signal; a high frequency regenerator (203, 213) to reconstruct a high band portion of the audio signal using the filtered low band audio signal and the high frequency reconstruction metadata, where the reconstruction includes a spectral translation if the parameter patchmode parameter is the first value and the reconstruction includes harmonic transposition by frequency expansion of the phase vocoder if the patchmode parameter is the second value; and a bank (203, 213) of synthesis filters for combining the filtered low-band audio signal with the regenerated high-band portion to form a wide-band audio signal; wherein the encoded audio bitstream further includes a padding element with an identifier indicating a start of the padding element and padding data after the identifier, wherein the padding data includes the backward compatible extension container, wherein the padding data include an extension payload, the extension payload includes spectral band replication extension data, and the extension payload is identified by a four-bit unsigned integer transmitted most significant bit first and having a value from '1101' or '1110'; wherein the high frequency reconstruction metadata includes envelope scale factors.