ES3030182T3 - Backward-compatible integration of high frequency reconstruction techniques for audio signals - Google Patents

Abstract

A method for decoding an encoded audio bitstream is described. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. It also includes extracting high-frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filter bank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether spectral translation or harmonic transposition should be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high-frequency reconstruction metadata according to the flag. (Automatic translation with Google Translate, no legal value)

Description

DESCRIPTION

Backward-compatible integration of high-frequency reconstruction techniques for audio signals

Cross-reference to the related application

This application is a divisional European application of European patent application EP 22189216.9 (reference: D17806EP04), for which EPO Form 1001 was filed on August 8, 2022.

Technical field

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 the 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 ISO/IEC 14496-3:2009 standard. In the MPEG-4 standard, AAC stands for “Advanced Audio Coding” and HE-AAC stands for “High Efficiency Advanced Audio Coding”.

The MPEG-4 AAC standard defines several audio profiles, which determine which coding objects and tools are present in a compliant 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 type 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.

The SBR object type contains the Spectral Band Replication tool, 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, enabling 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 through adaptive inverse filtering, as well as the optional addition of noise and sinusoidal noise. In the MPEG-4 AAC standard, the SBR tool performs spectral patching (also called linear translation or spectral translation), in which several consecutive Quadrature Mirror Filter (QMF) subbands are copied (or “patched”) from a transmitted low-band portion of an audio signal to a high-band portion of the audio signal, which is generated at 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 to improve spectral band replication are needed.

Document WO2016149015A1 relates to an audio processing unit including 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 beginning with an identifier followed by padding data. The padding data includes at least one flag identifying 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.

Brief description of embodiments of the invention

The present disclosure provides a method as detailed in claim 1. Also provided is a non-transitory computer-readable medium according to claim 6 and an audio processing unit according to claim 7. Advantageous features are provided in the dependent claims.

Brief description of the drawings

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

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

Figure 3 is a block diagram of a system including a decoder that is an embodiment of the audio processing unit of the invention and, optionally, also a post-processor 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 it is divided into.

Notation and Nomenclature

Throughout this description, including the claims, the expression performing an operation “on” a signal or data (for example, filtering, scaling, transforming, or applying gain to the signal or data) is used in a broad sense to mean 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 thereon).

Throughout this description, 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, pre-processing systems, post-processing systems, and bitstream processing systems (sometimes referred to as 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 description, including the claims, the term "couples" or "coupled" is used broadly to mean either a direct or indirect connection. Thus, if a first device is coupled to a second device, that connection may be through a direct connection, or through an indirect connection via 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 to 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 to be employed to decode the audio content of the bitstream. The term “SBR metadata” is used herein to indicate such metadata that is described or referenced in the MPEG-4 AAC standard for use with spectral band replication (“SBR”). As appreciated by one of skill in the art, 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 the SBR operates at the original sampling rate. The SBR encoder operates in parallel with the underlying core codec, albeit at a higher sampling rate. Although SBR is primarily a post-process in the decoder, important parameters are extracted in the encoder to ensure the most accurate high-frequency reconstruction at the decoder. The encoder estimates the spectral envelope of the SBR interval for a time and frequency interval/resolution appropriate to the characteristics of the current input signal segments. The spectral envelope is estimated using complex QMF analysis and subsequent energy calculation. The time and frequency resolutions of the spectral envelopes can be chosen with a high level of freedom to ensure the most appropriate time-frequency resolution for the given input segment. Envelope estimation must take into account that a transient in the original, mainly located in the high-frequency region (e.g., a treble zone), will be present to a lesser extent in the high band generated by the SBR before envelope adjustment, 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 on the time-frequency resolution of the spectral envelope data, compared to conventional spectral envelope estimation as used in other audio coding algorithms.

Apart from 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 low band constitutes a series of strong harmonics and the high band, which has to be recreated, consists primarily of random signal components, as well as situations where strong tonal components are present in the original high band with no counterparts in the low band, upon which the high band region is based. Furthermore, the SBR encoder works closely with the underlying core codec to assess which frequency range should be covered by the SBR at a given time. The SBR data is efficiently encoded before transmission by exploiting entropy coding, as well as 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 bit rate and sampling rate. This is because a lower bit rate generally implies a larger SBR interval 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 matching module. The system is based on a complex-valued QMF filter bank (for high-quality SBR) or a real-valued 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 envelope data from the bitstream. The underlying core decoder decodes the audio signal of the current frame (albeit at the lower 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 from the QMF filter bank. High frequencies are flexibly reconstructed from the low band based on given control parameters. Furthermore, the reconstructed high band is adaptively filtered on a subband-channel basis according to the control data to ensure appropriate spectral characteristics for 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 (referred to herein as a “block”) containing audio data (typically over a time period of 1024 or 960 samples) and related information and/or other data. The term “block” is used herein 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 (but not more than one) “raw_data_block” element.

Each block in an MPEG-4 AAC bitstream can include multiple syntactic elements (each of which is also realized in the bitstream as a data segment). Seven types of such syntactic elements are defined in the MPEG-4 AAC standard. Each syntactic element is identified by a different value of the "id_syn_ele" data element. Examples of syntactic elements include a "single_channel_element()", a "channel_pair_element()", and a "fill_element()". A single-channel element is a container that contains audio data from a single audio channel (a monophonic audio signal). A channel-pair element contains audio data from two audio channels (i.e., 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 mentioned above) followed by data, which is referred to as "padding data." Padding elements have historically been used to adjust the instantaneous bit rate of bit streams 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.

According to 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 decoder that receives bitstreams with padding data containing a new type of data may optionally be used by a device receiving the bitstream (e.g., a decoder) to extend the functionality of the device. Thus, as one of skill 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 (e.g., audio payloads containing channel data).

In some embodiments of the invention, the identifier used to identify a padding element may consist of a three-bit unsigned integer transmitted with the most significant bit (“uimsbf”) first having a value of 0x6. Multiple instances of the same type of syntactical element may occur within a block (e.g., multiple padding elements).

Another standard for encoding audio bitstreams is the MPEG Unified Audio and Speech Coding (USAC) standard (ISO/IEC 23003-3:2012). The MPEG USAC 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 extended and enhanced version of the SBR toolset 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).

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

An integer-order harmonic transponder T maps a sinusoid with frequency m to a sinusoid with frequency Tm, while preserving signal duration. Typically, three orders, T = 2, 3, 4, are used in sequence to produce each part of the desired output frequency interval using the smallest possible transposition order. If an output above the fourth-order transposition interval is required, it can be generated by frequency shifting. When possible, near-critically sampled baseband time domains are created for processing to minimize computational complexity.

The harmonic transponder can be based on either QMF or DFT. When using the QMF-based harmonic transponder, the bandwidth stretching of the core encoder's time-domain signal 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 transform stage. Since the QMF-based harmonic transponder does not feature adaptive frequency-domain oversampling of the signal, the corresponding flag in the bitstream (sbrOversamplingFlag[ch]) can be ignored.

When using the DFT-based harmonic transponder, 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 the core encoder samples coreCoderFrameLength), the nominal “full-size” transform size of the transponder is first determined by the signal adaptive frequency domain oversampling flag (sbrOversamplingFlag[ch]) in the bitstream.

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

preGain(k) = lO^^rg-towEnvsíopem/io^Q <k < k0

where k<0> is the first QMF subband in the master frequency band table and lowEnvSlope is calculated using a function that computes the coefficients of a best-fit polynomial (in a least-squares sense), such as polyfit(). For example,

po lyf i t(3,k0, xjowband, lowEnv, LowEnvSlope);

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

<»k(0,0)

lowEnv(k) = 10loglo ----- —— —------ — - ,0< k < k0

610numTimeSlots ■ RATE 60

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 time slot (e.g., 2), ^k is a linear prediction filter coefficient (potentially obtained from the covariance method), and where

A bitstream generated according to the MPEG USAC 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 to be applied by 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 to be employed to decode the audio content of the USAC bitstream.

In this specification, the term “enhanced SBR metadata” (or “eSBR metadata”) is used to denote metadata indicative of each type of spectral band replication processing to be applied by a decoder for decoding audio content from an encoded audio bitstream (e.g., a USAC bitstream) and/or that controls such spectral band replication processing, and/or is indicative of at least one feature or parameter of at least one SBR tool and/or eSBR tool to be employed for decoding such audio content, but which is not described or mentioned in the MPEG-4 AAC standard. An example of eSBR metadata is metadata (indicative of or for controlling spectral band replication processing) that is described or mentioned in the MPEG USAC standard but not in the MPEG-4 AAC standard. Therefore, eSBR metadata indicates herein metadata that is not SBR metadata, and SBR metadata indicates herein metadata that is not eSBR metadata.

A USAC bitstream may 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 (in accordance with the present invention) included in an MPEG-4 AAC bitstream (e.g., in the sbr_extension() container at the end of an SBR payload).

The eSBR processing performance, 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 the 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, and adds 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 (e.g., a small amount of control bits that are eSBR metadata is included) in one or more of the metadata segments of an encoded audio bitstream (e.g., 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 in 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 after the identifier.

Figure 1 is a block diagram of an exemplary 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 together as shown: encoder 1, delivery subsystem 2, decoder 3, and post-processing unit 4. In variations of the shown system, one or more of the elements are omitted, or additional audio data processing units are included.

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

One or more encoded audio bitstreams output from the encoder 1 may be acknowledged at the encoded audio delivery subsystem 2. The subsystem 2 is configured to store and/or deliver each encoded bitstream output from the encoder 1. An encoded audio bitstream output from the encoder 1 may be stored by the subsystem 2 (e.g., in the form of a DVD or Blu-ray disc), or transmitted by the subsystem 2 (which may implement a transmission or network link), or may be both stored and transmitted by the subsystem 2.

The decoder 3 is configured to decode an encoded MPEG-4 AAC audio bitstream (generated by the encoder 1) that it receives via the subsystem 2. In some embodiments, the 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., decoded PCM audio sample streams). 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 (e.g., decoded PCM audio sample streams). Typically, decoder 3 includes a buffer that stores (e.g., non-transiently) segments of the encoded audio bit stream received from subsystem 2.

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

Figure 2 is a block diagram of an encoder 100 that 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., ASICs, FPGAs, or other integrated circuits), in hardware, software, or a combination of hardware and software. The encoder 100 includes the encoder 105, the padding/formatting stage 107, the metadata generation stage 106, and the buffer 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 to an MPEG-4 AAC encoded output bitstream.

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

The encoder 105 is coupled and configured to encode (e.g., by 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 step 107.

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

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

Figure 3 is a block diagram of a system including a decoder 200 which is an embodiment of the audio processing unit of the invention, and optionally also a post-processor 300 coupled thereto. Any of the components or elements of the decoder 200 and the post-processor 300 may be implemented as one or more processes and/or one or more circuits (e.g., ASICs, FPGAs, or other integrated circuits), in hardware, software, or a combination of hardware and software. The decoder 200 comprises the buffer 201, the bitstream payload deformatter (parser) 205, the audio decoding subsystem 202 (sometimes referred to as the “core” decoding stage or “central” decoding subsystem), the eSBR processing stage 203, and the control bit generation stage 204, connected as shown. Typically also, the decoder 200 includes other processing elements (not shown).

The (buffer) memory 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 the buffer memory 201 is acknowledged to the deformatter 205.

In variations of the embodiment of Figure 3 (or the embodiment of Figure 4 to be described), a non-decoder APU (e.g., APU 500 of Figure 6) includes a buffer (e.g., a buffer 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 type received by buffer 201 of Figure 3 or Figure 4 (i.e., an encoded audio bitstream including 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 other metadata as well) therefrom, to commit at least the eSBR metadata and the SBR metadata to the eSBR processing stage 203, and typically also to commit 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 stage).

The system of Figure 3 optionally also includes the post-processor 300. The post-processor 300 includes a buffer 301 (buffer) 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 post-processor 300 from the decoder 200. The processing elements of the post-processor 300 are coupled and configured to receive and adaptively process a sequence of the blocks (or frames) of the decoded audio output from the buffer 301, using the metadata output of the decoding subsystem 202 (and/or deformatter 205) and/or control bits output from stage 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 commit the decoded audio data to the eSBR processing stage 203. The decoding is performed in the frequency domain and typically includes inverse quantization followed by spectral processing. Typically, a final processing stage in the subsystem 202 applies a frequency domain to time domain transformation to the decoded frequency domain audio data such 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 and eSBR metadata (extracted by parser 205) to the decoded audio data (i.e., to perform SBR and eSBR processing on the output of decoding subsystem 202 using the SBR and eSBR metadata) to generate the fully decoded audio data that is sent (e.g., to post-processor 300) from decoder 200. Typically, decoder 200 includes a memory (accessible by subsystem 202 and step 203) that stores the deformatted audio data and metadata output from deformatter 205, and step 203 is configured to access the audio data and metadata (including the SBR metadata and the eSBR metadata) to generate the fully decoded audio data that is sent (e.g., to post-processor 300) from decoder 200. Typically, decoder 200 includes a memory (accessible by subsystem 202 and step 203) that stores the deformatted audio data and metadata output from deformatter 205, and step 203 is configured to access the audio data and metadata (including the SBR metadata and the eSBR metadata) to generate the fully decoded audio data that is sent (e.g., to post-processor 300) from decoder 200. eSBR) as required during SBR and eSBR processing. The SBR processing and eSBR processing in step 203 can be considered post-processing on the output of the core 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) that is coupled and configured to upmix the output of stage 203 to generate fully decoded upmixed audio that is output from decoder 200. Alternatively, post-processor 300 is configured to upmix 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 acknowledged as output from the decoder 200 (e.g., to the post-processor 300 for use in post-processing). In response to the metadata extracted from the input bit stream (and optionally also in response to the control data), the stage 204 may generate (and acknowledge to the post-processor 300) control bits indicating that the decoded audio data output from the eSBR processing stage 203 is to undergo a specific type of post-processing. In some implementations, the decoder 200 is configured to commit metadata extracted by the deformatter 205 from the input bitstream to the post-processor 300, and the post-processor 300 is configured to perform post-processing on the decoded audio data output 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 inventive audio processing unit. The APU 210 is a legacy decoder that is not configured to perform eSBR processing. Any of the components or elements of the APU 210 may be implemented as one or more processes and/or one or more circuits (e.g., ASICs, FPGAs, or other integrated circuits), in hardware, software, or a combination of hardware and software. The APU 210 comprises buffer 201, bitstream payload deformatter 215 (parser), audio decoding subsystem 202 (sometimes referred to as the “core” decoding stage or “core” decoding subsystem), and SBR processing stage 213, connected as shown. Typically, the APU 210 also includes other processing elements (not shown). The APU 210 may represent, for example, an audio encoder, decoder, or transcoder.

Elements 201 and 202 of APU 210 are identical to the identically numbered elements of decoder 200 (of Figure 3) and the above description of them will not be repeated. In operation of APU 210, a sequence of blocks of an encoded audio bit stream (an MPEG-4 AAC bit stream) received by APU 210 is acknowledged from buffer 201 to 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 other metadata therefrom as well, but to ignore eSBR metadata that may be included in the bitstream in accordance with 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 in 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 commit the decoded audio data to the SBR processing stage 213. The decoding is performed in the frequency domain. Typically, a final processing stage in the subsystem 202 applies a frequency domain to time domain transformation to the decoded frequency domain audio data such 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 deformatter 215) to the decoded audio data (i.e., to perform SBR processing on the output of decoding subsystem 202 using the SBR metadata) to generate the fully decoded audio data that is output (e.g., to post-processor 300) from APU 210. Typically, APU 210 includes a memory (accessible by subsystem 202 and step 213) that stores the deformatted audio data and metadata output from deformatter 215, and step 213 is configured to access the audio data and metadata (including the SBR metadata) as needed during SBR processing. The SBR processing in stage 213 may be thought of as post-processing on the output of the core decoding subsystem 202. Optionally, the APU 210 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 the deformatter 215) that is coupled and configured to upmix the output of stage 213 to generate fully decoded, upmixed audio that is output from the APU 210. Alternatively, a post-processor is configured to upmix the output of the APU 210 (e.g., using PS metadata extracted by the deformatter 215 and/or control bits generated at 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 amount of control bits that are eSBR metadata are included) in an encoded audio bitstream (e.g., an MPEG-4 AAC bitstream), such that legacy decoders (that are not configured to parse the eSBR metadata, or to use any eSBR tools to which the eSBR metadata pertains) can ignore the eSBR metadata but nevertheless decode the bitstream as much as possible without using the eSBR metadata or any eSBR tools to which the eSBR metadata pertains, typically without any significant penalty in the quality of the decoded audio. However, eSBR decoders configured to analyze the bitstream to identify 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 for efficiently transmitting enhanced spectral band replication (eSBR) control data or metadata in a backward-compatible manner.

Typically, the eSBR metadata in the bitstream is indicative of (e.g., 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 may not have been applied by an encoder during 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 the values of the parameters (described in the USAC MPEG standard and in this description): sbrPatchingMode[ch], sbrOversamplingFlag[ch], sbrPitchInBins[ch], sbrPitchInBins[ch], and bs_sbr_preprocessing.

In this document, 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 expression [ch] and assume that the relevant parameter belongs to an audio content channel.

In this specification, the notation X[ch][env], where X is some parameter, indicates that the parameter belongs to the SBR envelope (“env”) of the audio content channel (“ch”) of an encoded bitstream to be decoded. For simplicity, the terms [env] and [ch] are sometimes omitted, and it is assumed that the relevant parameter belongs to an SBR envelope of an audio content channel.

During decoding of an encoded bitstream, the performance of harmonic transposition during an eSBR processing stage of 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 “sbrPatchingMode[ch]” value 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 either high-quality SBR 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 “sbrOversamplingFlag[ch]” value indicates the use of signal adaptive frequency domain oversampling in eSBR in combination with DFT-based SBR harmonic 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 is enabled as described in section 7.5.3.1 of the USAC MPEG standard; 0 indicates that signal adaptive frequency domain oversampling is disabled as described in section 7.5.3.1 of the USAC MPEG standard.

The “sbrPitchlnBinsFlag[ch]” value 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 “sbrPitchInBins[ch]” value controls the addition of cross-product terms in the SBR harmonic transponder. The sbrPitchinBins[ch] value is an integer in the range [0,127] and represents the measured distance in frequency bins for a 1536-line DFT acting on the core encoder sampling frequency.

In the case where an MPEG-4 AAC bitstream is indicative of an SBR channel pair 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_channel_pair_element().

The eSBR tool's harmonic transposition typically improves the quality of decoded musical signals at relatively low crossover frequencies. Non-harmonic transposition (i.e., legacy spectral patching) typically improves 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 based on the speech/music detection with harmonic transposition employed for musical content and spectral patching for velocity 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,” meaning that pre-flattening is performed or not depending on the value of this single bit. When using SBR’s QMF patching algorithm, as described in Section 4.6.18.6.3 of the MPEG-4 AAC standard, the pre-flattening stage may be performed (when directed by the “bs_sbr_preprocessing” parameter) in an effort to avoid discontinuities in the shape of the spectral envelope of a high-frequency signal being fed into 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 inclusion in the eSBR metadata of an MPEG-4 AAC bitstream indicative of the aforementioned eSBR tools (harmonic transposition and pre-flattening) is expected to be on the order of a few hundred bits per second because only the differential control data necessary to perform the eSBR processing is transmitted in accordance with some embodiments of the invention. Legacy decoders may ignore this information because it is included in a backward-compatible manner (as will be explained below). Therefore, the detrimental effect on bit rate associated with including eSBR metadata is negligible, for several reasons, including the following:

• The bit rate penalty (due to the inclusion of eSBR metadata) is a very small fraction of the total bit rate because only the differential control data needed to perform eSBR processing is transmitted (and not a simultaneous transmission of SBR control data); and

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

Thus, embodiments of the invention provide a means for efficiently transmitting 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 bit rate. Furthermore, the complexity and processing requirements associated with embodying eSBR according to embodiments of the invention are also reduced because the SBR data need only be processed once and not broadcast simultaneously, which would be the case if eSBR were treated 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.

7 , elements of a raw data block of an MPEG-4 AAC bitstream including eSBR metadata are described in accordance with some embodiments of the present invention. 7 is a diagram of a raw data block of the MPEG-4 AAC bitstream, showing segments thereof.

A block of an MPEG-4 AAC bitstream may include at least one “single_channel_element()” (e.g., 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 for an audio program. The block may also include a plurality of “fill_elements” (e.g., fill_element 1 and/or fill_element 2 in Figure 7) that include data (e.g., metadata) related to the program. Each “single_channel_element()” includes an identifier (e.g., “ID1” in Figure 7) indicating the start of a single-channel element, and may include audio data indicative of a different channel of a multi-channel audio program. Each channel_pair_element includes an identifier (not shown in Figure 7) indicating 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 filler element) of an MPEG-4 AAC bitstream includes an identifier (“ID2” in Figure 7) indicating the start of a filler element and filler data following the identifier. The ID2 identifier may consist of a three-bit unsigned integer most significant bit first (“uimsbf”) having a value of 0x6. The filler 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 most significant bit first (“uimsbf”).

The padding data (e.g., an extension payload thereof) may include a header or identifier (e.g., “header1” in Figure 7) indicating a padding data segment that is indicative of an SBR object (i.e., the header initializes an “SBR object” type, referred to as sbr_extension_data() in the MPEG-4 AAC standard). For example, a Spectral Band Replication (SBR) extension payload is identified by a value of ‘1101’ or ‘1110’ for the extension_type field in the header, with ‘1101’ identifying an extension payload containing SBR data, and ‘1110’ identifying an extension payload containing SBR data with a Cyclic Redundancy Check (CRC) to verify the correctness of the SBR data.

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

The MPEG-4 AAC standard provides 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 a padding element’s header (e.g., extension payload thereof) initializes an SBR object type (as does “header1” in Figure 7) and a Spectral Band Replication Extension Element of the padding element includes PS data, the padding element (e.g., extension payload thereof) includes spectral band replication data and a “bs_extension_id” parameter whose value (i.e., bs_extension_id = 2) indicates that PS data is included in a Spectral Band Replication Extension Element of the padding element.

According to some embodiments of the present invention, eSBR metadata (e.g., an indicator indicative of whether enhanced spectral band replication (eSBR) processing is to be performed on the audio content of the block) is included in a spectral band replication extension element of a padding element. For example, such an indicator is indicated in padding element 1 of Figure 7 , where the indicator appears after the header (the “SBR extension header” of padding element 1) of the “SBR extension element” of padding element 1. 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 (e.g., in the SBR extension element of padding 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 “bs_extension_id” parameter whose value (e.g., bs_extension_id = 3) indicates that eSBR metadata is included in the padding element and that eSBR processing should 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 (e.g., padding element 2 in Figure 7) of an MPEG-4 AAC bitstream other than a Spectral Band Replication Extension Element (SBR Extension Element) of the padding element. This is because padding elements containing an extension_payload() with SBR data or SBR data with a CRC do not contain any other extension payload 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” in Figure 7) indicating the start of a padding element and padding data following 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 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 enhanced spectral band 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 (e.g., header2 of Figure 7) has an identifying value that is not one of the conventional values specified in Table 4.57 of the MPEG-4 AAC standard, and instead is indicative of an eSBR extension payload (such that the extension_type field of the header indicates that the padding data includes eSBR metadata).

In a first class of embodiments, the invention is an audio processing unit (e.g., 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 bit stream (e.g., at least one block of an MPEG-4 AAC bit stream);

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 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, wherein 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 indicator that identifies whether enhanced spectral band replication (eSBR) processing is to be performed on the audio content of the block (e.g., using spectral band replication data and eSBR metadata included in the block).

The flag is the eSBR metadata, and an example of this flag is the sbrPatchingMode flag. Another example of this flag is the harmonicSBR flag. Both flags indicate whether a basic 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 filler 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 201 of Figure 4) that stores (e.g., non-transiently) at least said block of the encoded audio bit stream.

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 including eSBR metadata (indicative of these eSBR tools) is estimated to be as follows (for typical decoding with the given parameters):

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

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

° Based on QMF: 0.98 WMOPS;

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

It is known that DFT-based transposition typically performs better than QMF-based transposition 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 eSBR metadata is included in the padding element and that eSBR processing is to be 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 an sbr_extension() container of the padding element 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 sbr_extension() container of the filler element includes PS data, and such a parameter having the value bs_extension_id = 3 may indicate that a sbr_extension() container of the filler element includes eSBR metadata:

Table 1

According to some embodiments of the invention, the syntax of each spectral band replication extension element that includes eSBR metadata and/or PS data is as indicated in Table 2 below (where “sbr_extension()” indicates a container that is the spectral band replication extension element, “bs_extension_id” is as described in Table 1 above, “ps_data” indicates PS data, and “esbr_data” indicates eSBR metadata):

Table 2

In an exemplary 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 in 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 enhanced 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 either not already supported in the bitstream or can be inferred directly 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 already defined locations in the bitstream.

For example, an MPEG-4 HE-AAC or HE-AAC v2 compliant decoder may be extended to include an enhanced 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 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 AAC standard.

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

To improve subjective quality for audio content with strong harmonic frequency structure and 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 features, can be determined at the encoder by applying a signal-dependent classification mechanism. Generally, the use of the harmonic patching method (sbrPatchingMode == 1) is preferable for encoding musical 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. Conversely, the use of the regular SBR patching method is preferred for speech and mixed signals, as it provides better preservation of the temporal structure in speech.

To improve the performance of the harmonic transponder, a preprocessing stage (bs_sbr_preprocessing == 1) can be activated, which strives to avoid introducing spectral discontinuities into the signal entering the subsequent envelope adjuster. The tool's operation is beneficial for signal types where the approximate 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, adaptive frequency-domain oversampling can be applied (sbrOversamplingFlag == 1). Since adaptive frequency-domain oversampling increases the computational complexity of the transponder but is only beneficial for frames containing transients, the use of this tool is controlled by the bitstream element, which is transmitted once per frame and per separate 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 as long as the length of a central audio frame can be introduced, 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 according to embodiments of the invention. For example, envelope data and background noise data may 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 spectral band replication.

In essence, these embodiments leverage 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 amount of additional transmitted data possible. The metadata was originally tuned for a base form of HFR (e.g., SBR's spectral translation operation), but according to embodiments, is used for an enhanced form of HFR (e.g., eSBR's harmonic transposition). As discussed above, the metadata generally represents operational parameters (e.g., envelope scale factors, noise floor scale factors, time/frequency grid parameters, sinusoid addition information, variable crossover over frequency/band, inverse filtering mode, envelope resolution, smoothing mode, frequency interpolation mode) tuned for 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.

Therefore, extended decoders that support an enhanced form of spectral band replication can be created very efficiently by building on already defined bitstream elements (e.g., those in the SBR extension payload) and adding only the parameters needed to support the enhanced form of spectral band replication (in a padding element 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 enhanced form of spectral band replication by ensuring that the bitstream is backwards compatible with legacy decoders that do not support the enhanced form of spectral band replication. It will be appreciated that the reserved data field is a backwards compatible data field, i.e., it is a data field that is already supported by older 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 older 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 tool (eSBR) as indicated by 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 including a step of encoding audio data 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 the audio data) and uses the eSBR metadata to process the audio data to generate a decoded audio data stream.

Another aspect of the invention is an eSBR decoder configured to perform eSBR processing (e.g., using at least one of the known eSBR tools such 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), audio decoding subsystem 202 (sometimes referred to as the “core” decoding stage or “core” decoding subsystem, and which is identical to the decoding subsystem 202 of Figure 3), eSBR control data generation subsystem 401, and eSBR processing stage 203 (which is identical to stage 203 of Figure 3), connected as shown. Also typically, the decoder 400 includes other processing elements (not shown).

In operation of decoder 400, a sequence of blocks of an encoded audio bit stream (an MPEG-4 AAC bit stream) received by decoder 400 is acknowledged from buffer 201 to 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 other metadata therefrom as well. The deformatter 215 is configured to commit at least the SBR metadata to the eSBR processing stage 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 stage).

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 commit the decoded audio data to the eSBR processing stage 203. The decoding is performed in the frequency domain. Typically, a final processing stage in the subsystem 202 applies a frequency domain to time domain transformation to the decoded frequency domain audio data such 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 the subsystem 401, to the decoded audio data (i.e., to perform SBR and eSBR processing on 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) that stores the deformatted audio data and metadata output from the deformatter 215 (and optionally also from the subsystem 401), and step 203 is configured to access the audio data and metadata. as needed during SBR and eSBR processing. The SBR processing in stage 203 can be thought of as post-processing on the output of the core decoding subsystem 202. Optionally, the decoder 400 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 upmix the output of stage 203 to generate fully decoded, upmixed audio that is output from the APU 210.

Parametric stereo is a coding 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) inter-channel intensity differences (IIDs) that describe the intensity differences between channels; (2) inter-channel phase differences (IPDs) that describe the phase differences between channels; and (3) inter-channel coherence (ICC) that 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 for high-quality reconstruction of the stereo image. However, IPD parameters only specify the relative phase differences between channels of the input stereo signal and do not indicate the distribution of these phase differences across the left and right channels. Therefore, a fourth type of parameter that describes an overall phase shift or overall phase difference (OPD) can also be 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 together with spatial parameters to generate the reconstructed left (lk(n)) and right (rk(n)) signals according to:

lk(n)=Hn(k,n)Sk(n)+H<2>i(k,n)dk(n)

rk(n)=Hi<2>(k,n)Sk(n)+H2<2>(k,n)dk(n)

where H11, H12, H21 and H22 are defined by the stereo parameters. The signals lk(n) and rk(n) are finally transformed back to the time domain using a frequency-to-time transformation.

The control data generating 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 eSBR metadata of any of the types included in audio bitstreams encoded according to other embodiments of the invention) in response to at least one result of the detecting step. The eSBR control data is asserted in 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) for setting the sbrPatchingMode[ch] parameter (and asserting the set parameter 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 asserting the set parameter 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 for setting the sbrPitchInBinsFlag[ch] and sbrPitchlnBins[ch] parameters (and asserting the set parameters in step 203) in response to detecting the pitch of the audio content indicated by the bitstream. Other aspects of the invention are the methods of decoding audio bit streams performed by any embodiment of the decoder of the invention described in this paragraph and in the preceding 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) to perform. 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-transitory manner) 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 programmable general-purpose processor, 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, a 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 data committed thereto.

Embodiments of the present invention may be implemented in hardware, firmware, or software, or a combination thereof (e.g., as a programmable logic array). Unless otherwise specified, the algorithms or processes included as part of the invention are not intrinsically related to any particular computer or other apparatus. In particular, various general-purpose machines with programs written in accordance with the teachings herein may be used, or it may be more convenient to construct more specialized apparatus (e.g., integrated circuits) to perform the required steps of the method. Thus, the invention may be implemented in one or more computer programs executing on one or more programmable computer systems (e.g., an implementation of any of the elements of Figure 1, or the encoder 100 of Figure 2 (or an 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 output device or port. The program code is applied to the 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 either case, the language can be a compiled or interpreted language.

For example, when implemented by sequences of computer software instructions, various functions and steps of embodiments of the invention may be implemented by sequences of multi-threaded software instructions 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 such computer program is preferably stored or downloaded to a storage medium or device (e.g., solid-state memory or medium, or magnetic or optical medium) readable by a general-purpose or special-purpose programmable computer, so as to configure and operate the computer when the storage medium or device is read by the computer system to perform the methods 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, wherein the so-configured storage medium causes a computer system to operate in a specific, predefined manner to perform the functions described herein.

Various embodiments of the invention have been described. However, it will be understood that various modifications may 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 may be used in combination with complex QMF synthesis and analysis filter banks. The analysis filter bank is responsible for filtering the time-domain lowband signal generated by the core decoder into a plurality of subbands (e.g., QMF subbands). The synthesis filter bank is responsible for combining the regenerated highband produced by the selected HFR technique (as indicated by the received sbrPatchingMode parameter) with the decoded lowband to produce a wideband output audio signal. However, a given filter bank implementation operating in a certain sampling frequency mode, e.g., normal dual-rate operation or down-sampled SBR mode, should not, however, have phase shifts that are dependent on the bitstream. The QMF banks used in SBR are a complex exponential extension of the theory of cosine-modulated filter banks. It can be shown that aliasing cancellation restrictions become obsolete when the cosine-modulated filter bank is extended 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:

hk(ri)= /fc (n ) = p 0( r c ) e x p { i^ ( A :+ í ) ( r i - ^ ) } , 0 < n < N;0 < k < M(1)

where p<0>(n) is a real-valued symmetric or asymmetric prototype filter (typically, a low-pass prototype filter), M indicates 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 from 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 operating the synthesis filter bank in downsampling mode, the synthesis filter bank may have only 32 channels. Because the subband samples from the filter bank are complex-valued, a possibly channel-dependent additive phase-shift stage may be added to the analysis filter bank. These additional phase shifts must be compensated for before the synthesis filter bank. Although the phase-shift terms can in principle be of arbitrary values without destroying the operation of the QMF analysis/synthesis chain, they may 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, po(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 inferred from Table 4 by one or more mathematical operations such as rounding, subsampling, interpolation, and decimation.

Although the adjustment of the SBR-related control information is not typically dependent on the details of the transposition (as discussed above), in some embodiments, certain elements of the control data may be concurrently transmitted in the eSBR extension container (bs_extension_id==EXTENSION_ID_ESBR) to improve the quality of the regenerated signal. Some of the concurrently transmitted elements may include the noise floor data (e.g., noise floor scale factors and a parameter indicating the direction, in either the frequency or time direction, of delta encoding for each noise floor), the inverse filtering data (e.g., a parameter indicating the inverse filtering mode selected from one of no inverse filtering, a low level of inverse filtering, an intermediate level of inverse filtering, and a high level of inverse filtering), and the missing harmonics data (e.g., 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's transponder performed in the encoder and therefore, if appropriately adjusted 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 adjusted for the eSBR harmonic transponder. The additional bit rate required to transmit these two classes of metadata for the eSBR harmonic transponder is relatively low. Therefore, sending adjusted missing harmonics and/or inverse filtering control data in the eSBR extension container will increase the audio quality produced by the transponder while only minimally affecting the bit rate. To ensure backward compatibility with legacy decoders, the adjusted parameters for SBR spectral translation operation may 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 forms. Any 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 (1)

CLAIMS 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; extracting from the encoded audio bitstream the high-frequency reconstruction metadata, the high-frequency reconstruction metadata including operating parameters for a high-frequency reconstruction process, the operating parameters including a patching mode parameter located in a backward-compatible extension bin of the encoded audio bitstream, wherein 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 spreading, and the high-frequency reconstruction metadata including time/frequency grid information; filtering the decoded low-band audio signal to generate a filtered low-band audio signal; regenerating a high-band portion of the audio signal using the filtered low-band audio signal and the high-frequency reconstruction metadata, wherein the regeneration includes spectral translation if the patching mode parameter is the first value, and the regeneration includes harmonic transposition via phase vocoder frequency spreading if the patching mode parameter is the second value; combining the filtered low-band audio signal with the regenerated high-band portion to form a wideband audio signal; and wherein the encoded audio bit stream further includes a padding element with an identifier indicating a start of the padding element and padding data following the identifier, wherein the padding data includes the backward-compatible extension container, wherein the padding data includes 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 of '1101' or '1110'. 2. The method of claim 1, wherein the filtering is performed by an analysis filter bank including analysis filters, hk(n), which are modulated versions of a prototype filter, pü(n), according to: 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, wherein the prototype filter, p<0>(n), is inferred from the coefficients in the table: