WO2021203753A1 - 音频信号的增量编码方法及装置 - Google Patents

音频信号的增量编码方法及装置 Download PDF

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WO2021203753A1
WO2021203753A1 PCT/CN2020/140741 CN2020140741W WO2021203753A1 WO 2021203753 A1 WO2021203753 A1 WO 2021203753A1 CN 2020140741 W CN2020140741 W CN 2020140741W WO 2021203753 A1 WO2021203753 A1 WO 2021203753A1
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audio
data
auxiliary data
track
audio signal
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PCT/CN2020/140741
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English (en)
French (fr)
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黄旭
潘兴德
吴超刚
谭敏强
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全景声科技南京有限公司
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes

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  • the present disclosure relates to the technical field of digital audio production, and in particular to an incremental encoding method and device for audio signals.
  • panoramic sound also known as three-dimensional sound
  • panoramic sound is the most realistic way of presentation and expression of sound. Whether in nature, art or audiovisual entertainment, panoramic sound is the future development trend.
  • Panoramic sound is sometimes called three-dimensional sound, immersive sound, and panoramic sound signals are generally divided into audio data and auxiliary data.
  • Audio data can be mono or multi-channel signals, such as mono, stereo, 4.0 channels, 5.1 channels, 7.1 channels, 9.1 channels, 11.1 channels, 13.1 channels, 22.2 channels and the above sound Any combination of channel types, such as 7.1 channel signal + 4.0 channel signal + 6 stereo signals;
  • auxiliary data is generally used to define the spatial position or rendering method of audio data, which can improve the presentation effect of audio data, such as three-dimensional positioning information, It can make the audio more spatial and immersive, and the sound effects (such as equalizer, reverb, etc.) can process information, which can make the audio more diversified and enrich the auditory experience.
  • an audio data and its auxiliary data are collectively called a sound object, and audio data without auxiliary data is called a sound bed.
  • the typical panoramic sound technology that has been commercially available can refer to the national three-dimensional panoramic sound standard AVS2-P3 (GB/T 33475.3), the international standard MPEG-H (ISO/IEC 23008-3), Dolby Atmos and WANOS.
  • the audio data can be a mono signal, a stereo signal, a single-layer multi-channel signal, a multi-layer multi-channel signal (that is, a combination of multiple channel signals, distributed in different height planes), and the like.
  • some panoramic sound signals use a two-layer plane of the middle layer and the top layer (for example, 5.1.4 is a combination of 5.1 and 4.0 multi-channel audio signals, 5.1 is in the middle layer and 4.0 is on the top layer), and some panoramic sound signals use three layers. Layer plane and so on.
  • Some panoramic sound signals have only multi-layer audio data, but no auxiliary data, such as SMPTE's 22.2 three-dimensional sound system and AURO 9.1 system.
  • Some panoramic sound signals have both multi-layer multi-channel signals and auxiliary data, such as MPEG-H, Dolby Atmos and DTS:X systems.
  • the panoramic sound signal can also be all mono or stereo signals and auxiliary data.
  • the panoramic sound format like AAC, AC3, MP3 and other formats, is also a compressed audio format.
  • DAW Digital Audio Workstation
  • Pro Tools such as Pro Tools, Nuendo, Cubase, Logic Pro, Adobe Audition, etc.
  • These softwares are widely used in the production of movies and music, and can use professional audio plug-ins to create High-quality audio signal.
  • the second category is some audio and video application software, such as K song, short video, dubbing software and so on. These softwares are widely used in people's lives and change people's daily life and work in a subtle way.
  • This type of audio and video application software supports the editing and production of conventional audio formats (including PCM format, and currently commonly used compressed audio formats such as mp3, aac, wma, etc.), and with the curse of the Internet, you can upload, share, and watch your work anytime, anywhere Other people's works are highly entertaining and interactive.
  • each sound element (hereinafter referred to as the sound track) contained in it and its corresponding auxiliary data, which are respectively recorded as the sound track set C and the auxiliary data set E.
  • the auxiliary data corresponds to the audio track, and each audio track can contain 0, 1, or multiple auxiliary data.
  • the production process edits the existing audio track/auxiliary data by adding, deleting, replacing or any combination of the three methods; this step can be repeated, and the audio track set C'and auxiliary data are generated after completion Set E'.
  • a band can jointly produce a rock music in the dismantling and re-editing method described in the Chinese invention application with the application number 2020102093909, as shown in Figure 2.
  • the first person records the guitar track C1 and adds an equalizer E1 to it, then encodes C1 and E1 (the resulting compressed code stream is denoted as S0') and uploads; the second person decodes S0' and decodes the guitar it contains
  • the audio track and its equalizer are decoded, denoted as C1' and E1, and then input your own bass track C2 and add the reverb effect E2 to it, and then encode C1', E1, C2, and E2 (the generated compressed code stream Record it as S0”) and upload;
  • the third person solves C1”, E1, C2', E2, enters the keyboard track C3, and encodes it as S0”', and so on.
  • the existing audio codec technology needs to re-encode all the sound elements in the sound program.
  • This processing method requires higher coding complexity on the one hand, and on the other hand the sound quality (especially the sound quality of the unmodified part) will rapidly decrease with multiple encodings.
  • the first person needs to code C1, E1
  • the second person needs to code C1', E1, C2, E2
  • the third person needs to code C1", E1, C2', E2, C3, and so on.
  • the last person needs to encode the most data.
  • the guitar track C1 is the original sound recorded by the first person, the quality is the best, and has not been modified, but after the encoding, C1' is solved The quality will decrease, and the quality of C1" will be worse; for the same reason, the quality of the bass track C2' is also worse than C2.
  • the present disclosure provides a method and device for incremental encoding of audio signals. Its technical purpose is to distinguish between unmodified sound data and modified data based on the original audio signal, and only compress and encode the modified data.
  • the unmodified data is first parsed from the original code stream and its compressed data is organized into a new code stream with the compressed data generated by the modified data, that is, the modified part is incrementally encoded to avoid the loss of sound quality of the unmodified data and reduce the encoding the complexity.
  • An incremental encoding method for audio signals including:
  • T After adding, deleting or replacing the data in the T0 or editing in any combination of the three methods, T is obtained;
  • P4 Classify the T to obtain the unmodified audio signal T1 and the modified audio signal T2 included in the T;
  • P6 Encode the T2 to obtain an audio code stream S2;
  • P7 Multiplex the S1 and the S2 into a new audio code stream S'.
  • both the T0 and the T are composed of audio track data, or are composed of audio track data and auxiliary data.
  • the T1 includes only audio track data, or only auxiliary data, or includes audio track data and auxiliary data, or no data.
  • the step P6 only encodes the audio track data.
  • the step P6 only encodes the auxiliary data.
  • the step P6 encodes the audio track data and auxiliary data at the same time.
  • An incremental encoding device for audio signals including:
  • Audio signal input module input compressed audio signal S
  • the audio decoding module decodes the S to obtain a decoded audio signal T0, where the T0 includes an audio track data set A0 and an auxiliary data set B0;
  • the audio editing module after adding, deleting or replacing the data in the T0 or editing in any combination of the three methods, obtains T, and the T includes the audio track data set A1 and the set auxiliary data B1;
  • the audio classification module classifies the T, and obtains that the T includes an unmodified audio signal T1 and a modified audio signal T2;
  • the search module searches for the code stream field corresponding to the T1 in the S to be S1;
  • An audio encoding module which encodes the T2 to obtain an audio code stream S2;
  • the audio multiplexing module multiplexes the S1 and the S2 into a new audio code stream S'.
  • the audio editing module includes:
  • the audio track editing unit generates a new audio track set A1 after adding, deleting, replacing, or editing in any combination of the three methods on the audio track set A0;
  • the auxiliary data editing unit generates a new auxiliary data set B1 after adding, deleting, replacing, or editing in any combination of the three methods on the auxiliary data set B0.
  • the audio classification module includes:
  • the audio track classification unit divides the A1 into an unmodified part of the audio track and a modified part of the audio track;
  • the auxiliary data classification unit divides the B1 into auxiliary data of the unmodified part and auxiliary data of the modified part.
  • the audio signal input module inputs the compressed audio signal S; the audio decoding module decodes the S to obtain the decoded audio signal T0, and the T0 includes the audio track data set A0 and the auxiliary data set B0; audio
  • the editing module adds, deletes, or replaces the data in the T0 or edits in any combination of the three methods to obtain T.
  • the T includes the audio track data set A1 and the set auxiliary data B1; the audio classification module T is classified, and the T includes the unmodified audio signal T1 and the modified audio signal T2; the search module finds that the code stream field corresponding to the T1 in the S is S1; the audio encoding module performs The audio code stream S2 is obtained by encoding; the audio multiplexing module multiplexes the S1 and the S2 into a new audio code stream S'.
  • the unmodified data is first parsed from the original code stream to obtain its compressed data, and generated with the modified data
  • the compressed data is organized into a new code stream, that is, the modified part is incrementally coded to avoid the loss of sound quality of the unmodified data and reduce the coding complexity.
  • Figure 1 is a flow chart of a method for secondary production of an existing audio signal
  • Fig. 2 is a flowchart of a specific embodiment of the secondary production of an existing audio signal
  • Figure 3 is a flow chart of the method of the present invention.
  • Figure 4 is a schematic diagram of the device of the present invention.
  • Figure 5 is a schematic diagram of the specific implementation of the device of the present invention.
  • FIG. 6 is a flowchart of Embodiment 1 of the present invention.
  • FIG. 7 is a flowchart of Embodiment 2 of the present invention.
  • FIG. 8 is a flowchart of Embodiment 3 of the present invention.
  • the incremental encoding method for audio signals provided by the present invention includes the following steps:
  • decoding S to obtain the decoded audio signal T0; decoding S is to completely separate all the audio track data and auxiliary data contained in S (refer to the Chinese invention patent application with application number 2020102093909) to generate the original audio track data Set A0 and auxiliary data set B0;
  • T After editing the audio track data and its auxiliary data in T0 by adding, deleting or replacing or any combination of the three methods, T is obtained; this step can be repeated, and T is generated after the editing is completed, and T includes the audio track Data set A1 and auxiliary data set B1;
  • P4 Classify T, and get T including unmodified audio signal T1 and modified audio signal T2; that is, compare the data in T and T0 one by one, and mark the modified data and unmodified data separately;
  • P6 Encode the modified audio signal T2 to obtain the audio code stream S2;
  • P7 Multiplex S1 and the S2 into a new audio code stream S'.
  • FIG 4 is a schematic diagram of the incremental encoding device for audio signals according to the present invention.
  • the incremental encoding device includes an audio signal input module, an audio decoding module, an audio editing module, an audio classification module, a search module, Audio encoding module and audio multiplexing module.
  • Figure 5 is a schematic diagram of a specific embodiment of the incremental encoding device.
  • the audio editing module includes a track editing unit and an auxiliary data editing unit
  • the audio classification module includes a track classification unit and an auxiliary data classification unit.
  • Embodiment 1 Edit and produce the audio track in the existing audio signal, as shown in Figure 6, the specific steps are as follows:
  • 603 Edit and produce audio, and mark the changes of each audio track, including the following:
  • Delete audio track delete the n1 to n2 audio tracks, clear the track data of C[n1,...,n2]; mark P[n1,...,n2] as "delete” ( If it has been marked as "add”, the original mark will be overwritten); the value of k remains unchanged (although n1 to n2 are deleted, the track position still exists); 0 ⁇ n1 ⁇ n2 ⁇ k-1;
  • This step can be repeated;
  • the track is regarded as the modified part M[] (the track number is stored in the collection, the same below), the track corresponding to the "delete” mark is regarded as the modified part N[], and other elements in P are regarded as the unmodified part L[] ;
  • 605 Use the original audio signal S1 and the audio track mark set P[] to encode the produced audio track into a new audio signal.
  • Create a new empty code stream S2 first put the frame header of S1 into S2, and then scan each element i in P[] one by one: if P[i] ⁇ L[], then the i-th audio code in S1
  • the stream is directly put into S2 (from the 0th track, arranged in order, the same below); if P[i] ⁇ M[], then the audio track C[i] is encoded and put into S2; if P[i] ⁇ N[], do not perform any operation; after scanning, rearrange the track number and update the frame header.
  • the output S2 is the new audio stream; the total number of tracks in S2 is less than or equal to k.
  • Embodiment 2 Editing and producing auxiliary data in an existing audio signal, as shown in Fig. 7, the specific steps are as follows:
  • each track is denoted as E[0,...,k-1], which means that S1 contains k audio tracks;
  • the auxiliary data set is denoted as E [0,...,k-1][] (Because each track may contain auxiliary data, it is represented by a two-dimensional array, the same below), where each track is denoted as E[0][0 ,...,m0-1], E[1][0,...,m1-1],..., E[k-1][m k-1 -1], which means each track
  • the number of auxiliary data is m0, m1,..., m k-1 ; k ⁇ 0, m0, m1,..., m k-1 ⁇ 0;
  • Delete auxiliary data delete the n1i to n2i auxiliary data from the i-th track, and clear the auxiliary data data of E[i][n1i,...,n2i]; Q[i][n1i ,...,n2i] is marked as "delete” (if it has been marked as "add”, the original mark will be overwritten); the mi value remains unchanged (n1i,...,n2i are deleted, but the auxiliary data position still exists ); 0 ⁇ n1i ⁇ n2i ⁇ mi-1;
  • This step can be repeated;
  • the scanning After the scanning is completed, rearrange the auxiliary data number and update the frame header, and at the same time put the i-th audio track code stream field directly into the corresponding position of S2; after scanning all the k audio tracks, the output S2 at this time is New audio stream; the total number of auxiliary data of each audio track in S2 is less than or equal to mi.
  • Embodiment 3 Editing and producing the audio track and auxiliary data in the audio signal, as well as secondary/multiple production, as shown in Fig. 8, and the details are as follows:
  • Delete audio track delete the n1 to n2 audio tracks, clear the audio track data and auxiliary data of C[n1,...,n2]; mark P[n1,...,n2] as "Delete” (overwrite the original mark if it has been marked as "add”); the value of k remains unchanged (n1,...,n2 are deleted, but the track position still exists); 0 ⁇ n1 ⁇ n2 ⁇ k -1;
  • Delete auxiliary data delete the n5i to n6i auxiliary data from the i-th track, and clear the auxiliary data data of E[i][n5i,...,n6i]; Q[i][n5i ,...,n6i] is marked as "delete” (if it has been marked as "add”, the original mark will be overwritten); the mi value remains unchanged (although n5i to n6i are deleted, the auxiliary data position still exists); 0 ⁇ n5i ⁇ n6i ⁇ mi-1;
  • This step can be repeated;
  • auxiliary data data For auxiliary data data, scan each auxiliary data mark in Q[i][0,...,mi-1] (denoted as j): if Q[i][j] ⁇ L2[i] [], then put the j-th auxiliary data stream field attached to the i-th audio track stream in S1 directly into S2 (from the 0th auxiliary data, in order); if P[i][ j] ⁇ M2[i][], then encode the auxiliary data E[i][j] and put it into S2; if P[i][j] ⁇ N2[i][], no operation is performed.
  • the output S2 is the new audio code stream; the total number of audio tracks in S2 is ⁇ k, and the total number of auxiliary data for each audio track is less than or equal to mi .
  • step (806) If two/multiple productions are required, use the compressed audio signal S2 output in step (705) as the existing audio signal S1, start the next production process, repeat steps (701) to (706); after the production is completed , Output the final compressed audio stream.
  • the number of audio channels includes mono, stereo, 4.0, 5.1, 7.1, 9.1, 11.1, and 13.1. , 22.2 channels and any combination of the above-mentioned channel types; each audio signal can contain one or more audio tracks, and each audio track can contain 0, 1, or more auxiliary data.
  • Coding formats include conventional audio formats (such as MP3, AAC, AC3, etc.), panoramic sound audio formats (such as Atmos, WANOS, AVS, MPEG-H), etc.

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Abstract

一种音频信号的增量编码方法及装置，涉及数字音频制作技术领域。增量编码方法包括输入压缩音频信号S（P1）；对S进行解码得到T0（P2）；对T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T（P3）；对T进行分类，得到T中包括未修改的音频信号T1和修改的音频信号T2（P4）；查找T1在S中对应的码流字段S1 （P5）；对T2进行编码得到音频码流S2（P6）；将S1和S2复用成新的音频码流S'（P7）。由此区分未修改的声音数据和修改的数据，并仅对修改的数据做压缩编码，避免未修改数据的音质损失并降低编码复杂度。

Description

音频信号的增量编码方法及装置 技术领域
本公开涉及数字音频制作技术领域，尤其涉及一种音频信号的增量编码方法及装置。
背景技术
音频技术经过多年发展，立体声、5.1、7.1环绕声等系统已经获得了广泛的应用，但这些系统因缺乏声音的高度信息，最多只能呈现二维的声音。在真实的世界中，全景声(也称三维声)是声音最真实的呈现和表达方式，无论自然界、艺术领域或视听娱乐领域，全景声都是未来的发展趋势。
全景声有时也被称为三维声、沉浸声，全景声信号一般分为音频数据和辅助数据。音频数据可以是单声道或多声道信号，如单声道、立体声、4.0声道、5.1声道、7.1声道、9.1声道、11.1声道、13.1声道、22.2声道以及上述声道类型的任意组合，如7.1声道信号+4.0声道信号+6个立体声信号；辅助数据一般用于定义音频数据的空间位置或渲染方式，能够提升音频数据的呈现效果，比如三维定位信息，能使音频的空间感、沉浸感更强，以及音效(如均衡器、混响等)处理信息，能使音频更加多元化，丰富听觉体验。有时，也将一个音频数据及其辅助数据统一称为声音对象，将没有辅助数据的音频数据称为声床。目前已经商用的典型全景声技术可以参考三维全景声国家标准AVS2-P3(GB/T 33475.3)、国际标准MPEG-H(ISO/IEC 23008-3)、Dolby Atmos和WANOS等。
在全景声信号中，音频数据可以是单声道信号、立体声信号、单层多声道信号、多层多声道信号(即多个声道信号组合，分布在不同高度平面)等。例如，有些全景声信号采用中间层及顶层的两层平面(如5.1.4就是5.1和4.0两种多声道音频信号的组合，5.1在中间层，4.0在顶层)，有些全景声信号采用三层平面等。有些全景声信号只有多层音频数据，但没有辅助数据，例如SMPTE的22.2三维声系统和AURO 9.1系统等。有些全景声信号则既有多层多声道信号，也有辅助数据，例如MPEG-H、Dolby Atmos和DTS:X系统。当然，作为一个极端的例子，全景声信号也可以全部是单声道或立体声信号和辅助数据。
全景声音格式和AAC、AC3、MP3等格式一样，也属于压缩音频格式。目前在制作压缩音频信号时普遍采用两类制作工具。第一类是数字音频工作站(Digital Audio Workstation，DAW，比如Pro Tools、Nuendo、Cubase、Logic Pro、Adobe Audition等)，这些软件广泛应用于电影和音乐的制作，能够使用专业的音频插件，制作出高质量的音频信号。
第二类是一些音视频应用软件，如K歌、短视频、配音软件等等。这些软件广泛深入大众生活，以潜移默化的方式改变着人们的日常生活和工作。这类音视频应用软件支持常规音频格式(包括PCM格式，以及mp3、aac、wma等目前常用的压缩音频格式)的编辑制作，并在互联网的加持下，能够随时随地上传分享自己的作品以及观看其他人的作品，具有很强的娱乐性和互动性。
随着音频制作的日益普及，制作方式也变得五花八门，比如在已有音频信号基础上直接进行二次制作。在互联网应用中，多人可以用接力的方 式共同完成一部作品(如多人配音、合唱、合奏等)，每个人在前一个人的作品(即已有压缩音频信号)基础上进行编辑制作，把自己的制作成果融入作品中，然后传给下一个人继续制作。以目前的技术，在已有音频信号基础上进行二次制作的方法如图1所示(参考申请号为2020102093909中国发明申请)，包括以下步骤：
(101)导入已有音频信号S0，并将其包含的每个声音元素(以下简称音轨)及其对应的辅助数据解出，分别记作音轨集合C和辅助数据集合E。辅助数据和音轨对应，每个音轨可包含0个、1个或多个辅助数据。
(102)进行编辑制作，制作过程通过添加、删除、替换或三种方式的任意组合对已有音轨/辅助数据进行编辑；此步骤可重复进行，完成后生成音轨集合C'和辅助数据集合E'。
(103)将音轨集合C'和辅助数据集合E'编码成新的压缩音频信号S0'。
例如，一个乐队按照申请号为2020102093909的中国发明申请所述的可拆解和再编辑的方式，共同制作一首摇滚乐，如图2所示。第一个人录入吉他音轨C1并为其添加均衡器E1，然后将C1和E1编码(生成的压缩码流记作S0')并上传；第二个人将S0'解码，将其包含的吉他音轨及其均衡器解出，记作C1'和E1，然后录入自己的贝斯音轨C2并为其添加混响效果E2，然后将C1'、E1、C2、E2编码(生成的压缩码流记作S0”)并上传；第三个人解出C1”、E1、C2'、E2，录入键盘音轨C3，编码成S0”'，以此类推。
然而，现有的音频编解码技术需要对声音节目中的所有声音元素重新编码。这种处理方式，一方面需要较高编码的复杂度，另一方面是声音质量(尤指未修改部分的声音质量)会随着多次编码而快速下降。上例中， 第一个人需要将C1、E1编码，第二个人需要将C1'、E1、C2、E2编码，第三个人需要将C1”、E1、C2'、E2、C3编码，以此类推，最后一个人需要编码的数据是最多的。另外，吉他音轨C1是第一个人录入的原声，质量最好，且始终未被修改过，但经过编码之后，再解出的C1'质量就会下降，C1”质量更差；同理，贝斯音轨C2'质量也比C2差。
发明内容
本公开提供了一种音频信号的增量编码方法及装置，其技术目的是：在原有音频信号的基础上，区分未修改的声音数据和修改的数据，并仅对修改的数据做压缩编码，未修改的数据首先从原始码流中解析出其压缩数据，并和修改数据生成的压缩数据组织成新的码流，即对修改部分做增量编码，避免未修改数据的音质损失并降低编码复杂度。
本公开的上述技术目的是通过以下技术方案得以实现的：
一种音频信号的增量编码方法，包括：
P1：输入压缩音频信号S；
P2：对所述S进行解码得到解码后的音频信号T0；
P3:对所述T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T；
P4：对所述T进行分类，得到所述T中包括的未修改的音频信号T1和修改的音频信号T2；
P5：查找所述T1在所述S中对应的码流字段S1；
P6：对所述T2进行编码得到音频码流S2；
P7：将所述S1和所述S2复用成新的音频码流S'。
进一步地，所述T0和所述T均由音轨数据组成，或由音轨数据和辅助数据共同组成。
进一步地，所述T1仅包括音轨数据，或仅包括辅助数据，或包括音轨数据和辅助数据，或无任何数据。
进一步地，若所述T2仅包含音轨数据，则所述步骤P6仅对音轨数据进行编码。
进一步地，若所述T2仅包含辅助数据，则所述步骤P6仅对辅助数据进行编码。
进一步地，若所述T2包含音轨数据和辅助数据，则所述步骤P6对音轨数据和辅助数据同时进行编码。
一种音频信号的增量编码装置，包括：
音频信号输入模块,输入压缩音频信号S；
音频解码模块，对所述S进行解码得到解码后的音频信号T0，所述T0包括音轨数据集合A0和辅助数据集合B0；
音频编辑模块，对所述T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T，所述T包括音轨数据集合A1和集合辅助数据B1；
音频分类模块，对所述T进行分类，得到所述T中包括未修改的音频信号T1和修改的音频信号T2；
查找模块，查找所述T1在所述S中对应的码流字段为S1；
音频编码模块，对所述T2进行编码得到音频码流S2；
音频复用模块，将所述S1和所述S2复用成新的音频码流S'。
进一步地，所述音频编辑模块包括：
音轨编辑单元，对所述音轨集合A0进行添加、删除或替换或三种方式的任意组合的编辑后，生成新的音轨集合A1；
辅助数据编辑单元，对所述辅助数据集合B0进行添加、删除或替换或三种方式的任意组合的编辑后，生成新的辅助数据集合B1。
进一步地，所述音频分类模块包括：
音轨分类单元，将所述A1分为未修改部分的音轨和修改部分的音轨；
辅助数据分类单元，将所述B1分为未修改部分的辅助数据和修改部分的辅助数据。
本公开的有益效果在于：音频信号输入模块输入压缩音频信号S；音频解码模块对所述S进行解码得到解码后的音频信号T0，所述T0包括音轨数据集合A0和辅助数据集合B0；音频编辑模块对所述T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T，所述T包括音轨数据集合A1和集合辅助数据B1；音频分类模块对所述T进行分类，得到所述T中包括未修改的音频信号T1和修改的音频信号T2；查找模块查找所述T1在所述S中对应的码流字段为S1；音频编码模块对所述T2进行编码得到音频码流S2；音频复用模块将所述S1和所述S2复用成新的音频码流S'。
在原有音频信号的基础上，区分未修改的声音数据和修改的数据，并仅对修改的数据做压缩编码，未修改的数据首先从原始码流中解析出其压缩数据，并和修改数据生成的压缩数据组织成新的码流，即对修改部分做增量编码，避免未修改数据的音质损失并降低编码复杂度。
附图说明
图1为现有音频信号二次制作的方法流程图；
图2为现有音频信号二次制作的具体实施例流程图；
图3为本发明方法流程图；
图4为本发明装置示意图；
图5为本发明装置具体实施的示意图；
图6为本发明实施例一流程图；
图7为本发明实施例二流程图；
图8为本发明实施例三流程图。
具体实施方式
下面将结合附图对本公开技术方案进行详细说明。
本发明提供的音频信号的增量编码方法，如图3所示，包括如下步骤：
P1：输入压缩音频信号S；
P2：对S进行解码得到解码后的音频信号T0；对S进行解码即将S中包含的所有音轨数据和辅助数据完全分离(参考申请号为2020102093909的中国发明申请专利)，生成原始音轨数据集合A0和辅助数据集合B0；
P3:对T0中的音轨数据及其辅助数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T；此步骤可反复进行，编辑完成后生成T，T则包括音轨数据集合A1和辅助数据集合B1；
P4：对T进行分类，得到T中包括未修改的音频信号T1和修改的音频信号T2；即将T和T0中的数据进行逐一比对，将修改的数据和未修改的数据分别标记出来；
P5：在原有的压缩音频信号S中，找到未修改的音频信号T1对应的码 流字段S1并保留；
P6：对修改的音频信号T2进行编码得到音频码流S2；
P7：将S1和所述S2复用成新的音频码流S'。
图4为本发明所述的音频信号的增量编码装置的示意图，如图4所示，该增量编码装置包括音频信号输入模块、音频解码模块、音频编辑模块、音频分类模块、查找模块、音频编码模块和音频复用模块。图5为增量编码装置具体实施例的示意图，由图5可知，音频编辑模块包括音轨编辑单元和辅助数据编辑单元，音频分类模块包括音轨分类单元和辅助数据分类单元。
实施例一：对已有音频信号中的音轨进行编辑制作，如图6所示，具体步骤如下：
601：导入已有音频信号，记作S1；
602：将S1解码，得到音轨集合，记作C[0,...,k-1]，表示S1中包含k个音轨，k≥0；
603：对音频进行编辑制作，同时将每个音轨的改动情况进行标记,包含如下情况：
(1)添加音轨:将添加的音轨数量记作k1，并将添加的音轨放在C[k,k+1,...,k+k1-1]中，即目前音轨共有k+k1个；同时设立标记集合P[0,...,k+k1-1]，将P[k,...,k+k1-1]标记为“添加”；更新k值，使其始终等于当前音轨总数，即k＝k+k1,k1≥0；
(2)删除音轨:删除第n1至n2个音轨，将C[n1,...,n2]的音轨数据清空；将P[n1,...,n2]标记为“删除”(如果已标记为“添加”则覆盖原 有标记)；k值保持不变(n1至n2虽然被删除，但音轨位置依然存在)；0≤n1≤n2≤k-1；
(3)替换音轨：替换第n3至n4个音轨，则C[n3,...,n4]的音轨数据发生变化，将P[n3,...,n4]标记为“替换”(如果已标记为“添加”则覆盖原有标记)，k值保持不变；0≤n3≤n4≤k-1；
此步骤可重复进行；
604：将制作前后的音轨进行逐一对比，此时音轨总数为k，则将标记集合P[0,...,k-1]中的所有“添加”、“替换”标记对应的音轨视为修改部分M[](集合中储存的是音轨编号，下同)，“删除”标记对应的音轨视为修改部分N[]，P中其他元素视为未修改部分L[]；
605：利用原始音频信号S1、音轨标记集合P[]，将制作后的音轨编码成新的音频信号。新建空码流S2，先将S1的帧头放入S2中，然后逐个扫描P[]中的每个元素i：若P[i]∈L[]，则将S1中的第i个音频码流直接放入S2中(从第0个音轨起，按顺序依次排放，下同)；若P[i]∈M[]，则将音轨C[i]编码，放入S2中；若P[i]∈N[]，则不进行任何操作；扫描完成后，重新整理音轨编号并更新帧头，此时输出的S2即为新的音频码流；S2中的音轨总数小于等于k。
实施例二：对已有音频信号中的辅助数据进行编辑制作，如图7所示，具体步骤如下：
(701)导入已有音频信号，记作S1；
(702)将S1解码，得到音轨和辅助数据集合，其中音轨集合记作C[0,...,k-1]，表示S1中包含k个音轨；将辅助数据集合记作 E[0,...,k-1][](由于每个音轨都可能包含辅助数据，故用二维数组表示，下同)，其中每个音轨分别记作E[0][0,...,m0-1]、E[1][0,...,m1-1]、...、E[k-1][m k-1-1]，表示每个音轨的辅助数据数量分别是m0、m1、...、m k-1；k≥0，m0、m1、...、m k-1≥0；
(703)对音频进行编辑制作，同时将每个辅助数据的改动情况进行标记，包含如下情况：
(1)添加辅助数据：对第i个音轨添加辅助数据，将添加的辅助数据数量记作ni，并将添加的辅助数据放在E[i][mi,...,mi+ni-1]，即目前第i个音轨共有mi+ni个辅助数据；同时为每个音轨设立辅助数据标记集合Q[i][0,...,mi+ni-1]，将Q[i][mi,...,mi+ni-1]标记为“添加”；更新mi值，使其始终等于第i个音轨总数，即mi＝mi+ni；0≤i≤k-1，ni≥0；
(2)删除辅助数据:从第i个音轨上删除第n1i至n2i个辅助数据，将E[i][n1i,...,n2i]的辅助数据数据清空；将Q[i][n1i,...,n2i]标记为“删除”(如果已标记为“添加”则覆盖原有标记)；mi值保持不变(n1i,...,n2i虽然被删除，但辅助数据位置依然存在)；0≤n1i≤n2i≤mi-1；
(3)替换辅助数据:在第i个音轨上替换第n3i至n4i个辅助数据，则E[i][n3i,...,n4i]的辅助数据数据发生变化，将Q[i][n3i,...,n4i]标记为“替换”(如果已标记为“添加”则覆盖原有标记)，mi值保持不变；0≤n3i≤n4i≤mi-1；
此步骤可重复进行；
(704)对于每个音轨，将制作前后的辅助数据进行逐一对比。此时每 个音轨上的辅助数据总数为mi，则将标记集合Q[i][0,...,mi-1]中的所有“添加”“替换”标记对应的辅助数据视为修改部分M[i][](集合中储存的是辅助数据编号，下同)，“删除”标记对应的音轨视为修改部分N[i][]，Q中其他元素视为未修改部分L[i][]；
(705)利用原始音频信号S1、辅助数据标记集合Q[][]，将制作后的音轨和辅助数据编码成新的音频信号。新建空码流S2，先将S1的帧头放入S2中，然后对每个音轨逐个扫描Q[i][0,...,mi-1]中的每个辅助数据标记(记作j)：若Q[i][j]∈L[i][]，则将S1中第i个音轨码流附属的第j个辅助数据码流字段直接放入S2中(从第0个音轨起，按顺序依次排放；对于每个音轨，从第0个辅助数据起，按顺序依次排放，下同)；若Q[i][j]∈M[i][]，则将辅助数据E[i][j]编码，放入S2中；若Q[i][j]∈N[i][]，则不进行任何操作。扫描完成后，重新整理辅助数据编号并更新帧头，同时将第i个音轨码流字段直接放入S2的对应位置中；将k个音轨全部扫描完成后，此时输出的S2即为新的音频码流；S2中每个音轨的辅助数据总数小于等于mi。
实施例三：对音频信号中的音轨和辅助数据进行编辑制作以及二次/多次制作，如图8所示，具体如下：
(801)导入已有音频信号，记作S1；
(802)将S1解码，得到音轨和辅助数据集合，其中音轨集合记作C[0,...,k-1]，表示S1中包含k个音轨；将辅助数据集合记作E[0,...,k-1][]，其中每个音轨分别记作E[0][0,...,m0-1]、E[1][0,...,m1-1]、...、E[k-1][m k-1-1]，表示每个音轨的辅助数据数量 分别是m0、m1、...、m k-1；k≥0，m0、m1、...、m k-1≥0；
(803)对音频进行编辑制作，包含如下情况：
(1)添加音轨：将添加的音轨数量记作k1，并将添加的音轨放在C[k,k+1,...,k+k1-1]，即目前音轨共有k+k1个；同时设立标记集合P[0,...,k+k1-1]，将P[k,...,k+k1-1]标记为“添加”；更新k值，使其始终等于当前音轨总数，即k＝k+k1；k1≥0；
(2)删除音轨：删除第n1至n2个音轨，将C[n1,...,n2]的音轨数据及其辅助数据清空；将P[n1,...,n2]标记为“删除”(如果已标记为“添加”则覆盖原有标记)；k值保持不变(n1,...,n2虽然被删除，但音轨位置依然存在)；0≤n1≤n2≤k-1；
(3)替换音轨：替换第n3至n4个音轨，则C[n3,...,n4]的音轨数据发生变化，将P[n3,...,n4]标记为“替换”(如果已标记为“添加”则覆盖原有标记)，k值保持不变；0≤n3≤n4≤k-1；
(4)添加辅助数据：对第i个音轨添加辅助数据，将添加的辅助数据数量记作ni，并将添加的辅助数据放在E[i][mi,...,mi+ni-1]，即目前第i个音轨共有mi+ni个辅助数据；同时为每个音轨设立辅助数据标记集合Q[i][0,...,mi+ni-1]，将Q[i][mi,...,mi+ni-1]标记为“添加”；更新mi值，使其始终等于第i个音轨总数，即mi＝mi+ni；0≤i≤k-1，ni≥0；
(5)删除辅助数据：从第i个音轨上删除第n5i至n6i个辅助数据，将E[i][n5i,...,n6i]的辅助数据数据清空；将Q[i][n5i,...,n6i]标记为“删除”(如果已标记为“添加”则覆盖原有标记)；mi值保持不变(n5i至n6i虽然被删除，但辅助数据位置依然存在)；0≤n5i≤n6i≤mi-1；
(6)替换辅助数据：在第i个音轨上替换第n7i至n8i个辅助数据，则E[i][n7i,...,n8i]的辅助数据数据发生变化，将Q[i][n7i,...,n8i]标记为“替换”(如果已标记为“添加”则覆盖原有标记)，mi值保持不变；0≤n7i≤n8i≤mi-1；
此步骤可重复进行；
(804)将制作前后的音轨和辅助数据进行对比：此时音轨总数为k，则将标记集合P[0,...,k-1]中的所有“添加”“替换”标记对应的音轨视为修改部分M1[](集合中储存的是音轨编号，下同)，“删除”标记对应的音轨视为修改部分N1[]，P中其他元素视为未修改部分L1[]；此时每个音轨上的辅助数据总数为mi，则将标记集合Q[i][0,...,mi-1]中的所有“添加”“替换”标记对应的辅助数据视为修改部分M2[i][](集合中储存的是辅助数据编号，下同)，“删除”标记对应的音轨视为修改部分N2[i][]，Q中其他元素视为未修改部分L2[i][]；
(805)利用原始音频信号S1、音轨标记集合P[]、辅助数据标记集合Q[][]，将制作后的音轨和辅助数据编码成新的音频信号。
新建空码流S2，先将S1的帧头放入S2中，然后按照音轨逐个扫描：
(1)对于音轨数据，逐个扫描P[]中的每个元素i：若P[i]∈L1[]，则将S1中的第i个音频码流直接放入S2中(从第0个音轨起，按顺序依次排放)；若P[i]∈M1[]，则将音轨C[i]编码，放入S2中；若P[i]∈N1[]，则不进行任何操作；
(2)对于辅助数据数据，扫描Q[i][0,...,mi-1]中的每个辅助数据标记(记作j)：若Q[i][j]∈L2[i][]，则将S1中第i个音轨码流附属的第j 个辅助数据码流字段直接放入S2中(从第0个辅助数据起，按顺序依次排放)；若P[i][j]∈M2[i][]，则将辅助数据E[i][j]编码，放入S2中；若P[i][j]∈N2[i][]，则不进行任何操作。
扫描完成后，重新整理音轨和辅助数据编号并更新帧头，此时输出的S2即为新的音频码流；S2中的音轨总数≤k，每个音轨的辅助数据总数小于等于mi。
(806)若需要二次/多次制作，则将步骤(705)输出的压缩音频信号S2作为已有音频信号S1，开始下一次制作过程，重复步骤(701)至(706)；制作完毕后，输出最终的压缩音频流。
作为具体实施例地，上述处理过程中描述的所有音频信号，音频声道数包括单声道、立体声、4.0声道、5.1声道、7.1声道、9.1声道、11.1声道、13.1声道、22.2声道以及上述声道种类的任意组合形式；每个音频信号均可包含一个或多个音轨，每个音轨都可包含0个、1个或多个辅助数据。编码格式包括常规音频格式(如MP3、AAC、AC3等)、全景声音频格式(如Atmos、WANOS、AVS、MPEG-H)等。
以上为本公开示范性实施例，本公开的保护范围由权利要求书及其等效物限定。

Claims (9)

  1. 一种音频信号的增量编码方法，其特征在于，包括：
    P1：输入压缩音频信号S；
    P2：对所述S进行解码得到解码后的音频信号T0；
    P3:对所述T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T；
    P4：对所述T进行分类，得到所述T中包括未修改的音频信号T1和修改的音频信号T2；
    P5：查找所述T1在所述S中对应的码流字段S1；
    P6：对所述T2进行编码得到音频码流S2；
    P7：将所述S1和所述S2复用成新的音频码流S'。
  2. 如权利要求1所述的音频信号的增量编码方法，其特征在于，所述T0和所述T均由音轨数据组成，或由音轨数据和辅助数据共同组成。
  3. 如权利要求2所述的音频信号的增量编码方法，其特征在于，所述T1仅包括音轨数据，或仅包括辅助数据，或包括音轨数据和辅助数据，或无任何数据。
  4. 如权利要求3所述的音频信号的增量编码方法，其特征在于，若所述T2仅包含音轨数据，则所述步骤P6仅对音轨数据进行编码。
  5. 如权利要求3所述的音频信号的增量编码方法，其特征在于，若所述T2仅包含辅助数据，则所述步骤P6仅对辅助数据进行编码。
  6. 如权利要求3所述的音频信号的增量编码方法，其特征在于，若所述T2包含音轨数据和辅助数据，则所述步骤P6对音轨数据和辅助数据同时进行编码。
  7. 一种音频信号的增量编码装置，其特征在于，包括：
    音频信号输入模块,输入压缩音频信号S；
    音频解码模块，对所述S进行解码得到解码后的音频信号T0，所述T0包括音轨数据集合A0和辅助数据集合B0；
    音频编辑模块，对所述T0中的数据进行添加、删除或替换或三种方式的任意组合的编辑后，得到T，所述T包括音轨数据集合A1和集合辅助数据B1；
    音频分类模块，对所述T进行分类，得到所述T中包括未修改的音频信号T1和修改的音频信号T2；
    查找模块，查找所述T1在所述S中对应的码流字段为S1；
    音频编码模块，对所述T2进行编码得到音频码流S2；
    音频复用模块，将所述S1和所述S2复用成新的音频码流S'。
  8. 如权利要求7所述的音频信号的增量编码装置，其特征在于，所述音频编辑模块包括：
    音轨编辑单元，对所述音轨集合A0进行添加、删除或替换或三种方式的任意组合的编辑后，生成新的音轨集合A1；
    辅助数据编辑单元，对所述辅助数据集合B0进行添加、删除或替换或三种方式的任意组合的编辑后，生成新的辅助数据集合B1。
  9. 如权利要求8所述的音频信号的增量编码装置，其特征在于，所述音频分类模块包括：
    音轨分类单元，将所述A1分为未修改部分的音轨和修改部分的音轨；
    辅助数据分类单元，将所述B1分为未修改部分的辅助数据和修改部分 的辅助数据。
PCT/CN2020/140741 2020-04-10 2020-12-29 音频信号的增量编码方法及装置 WO2021203753A1 (zh)

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