CN112953934A

CN112953934A - DAB low-delay real-time voice broadcasting method and system

Info

Publication number: CN112953934A
Application number: CN202110184365.4A
Authority: CN
Inventors: 杨虹; 朱金晨; 张红升; 费林坤; 刘红江
Original assignee: Chongqing University of Post and Telecommunications
Current assignee: Chongqing University of Post and Telecommunications
Priority date: 2021-02-08
Filing date: 2021-02-08
Publication date: 2021-06-11
Anticipated expiration: 2041-02-08
Also published as: CN112953934B

Abstract

The invention relates to a method for DAB low-delay real-time voice broadcasting, which belongs to the technical field of digital signal broadcasting and comprises the following steps: s1: collecting audio; s2: coding by adopting EVS with code rate of 9.6kbps and bandwidth of NB, WB or SWB mode, and encapsulating each EVS frame into FIBs of 1 DAB; s3: synthesizing DAB transmission frames by adopting a DAB mode III, and distributing continuous 6 FIBs containing EVS frames in 5 DAB transmission frames according to a certain rule for transmission; s4: and receiving the DAB frame, decoding by adopting the EVS, and playing the audio. And to a DAB low-delay real-time voice broadcasting system. The invention avoids time interleaving delay existing in audio transmission through a DAB main service channel and greatly shortens coding and decoding time, the integral delay of an actual system does not exceed 100ms and is lower than the delay which can be sensed by human ears, thereby realizing the low-delay real-time voice broadcast based on the DAB.

Description

DAB low-delay real-time voice broadcasting method and system

Technical Field

The invention belongs to the technical field of digital signal broadcasting, and relates to a method and a system for DAB low-delay real-time voice broadcasting.

Background

Digital Audio Broadcasting (DAB) is a new Broadcasting system that gradually replaces the traditional fm Broadcasting with the continuous development of Broadcasting technology, and is a product of the development of Digital technology. Compared with the traditional broadcasting system, the DAB has more excellent listening tone quality and anti-interference capability, and also has a series of advantages of small transmitting power, large coverage area, high frequency spectrum utilization rate, movable reception and the like. DAB can transmit high-quality digitized sounds and at the same time can transmit pictures, texts, videos, etc. compared to conventional analog radio broadcasting. Compared with wireless technologies such as WIFI and Bluetooth, the DAB transmission distance is far, and wireless transmission of hundreds of meters and even kilometers can be realized, so that the system is more suitable for high-quality audio broadcasting in scenes such as schools, scenic spots and squares.

The DAB system can be used not only for city-level commercial broadcasting, but also for broadcasting and information distribution in small areas such as schools, scenic spots and the like, wherein an important application scene is live real-time voice broadcasting. At present, the mainstream DAB real-time voice broadcasting system mainly comprises audio acquisition, audio coding, DAB transmission frame synthesis, channel coding, OFDM modulation, up-conversion and other parts, and after a receiving end performs operations such as demodulation and decoding, a user can hear digitized audio. However, because the DAB system implements time interleaving with an interleaving depth of 384ms in a Main traffic Channel (MSC), and the format of the DAB audio frame is MP2 or LC AAC, a large amount of compression and decoding time is required (each frame is 24ms, and each compression and decoding requires at least 2 frame buffers, so at least 96ms is required), and the time required for Channel decoding is 2-24 ms, which causes a delay of at least 482-504 ms in DAB audio transmission. The DAB system is difficult to be used for real-time voice broadcasting in a small area due to a large transmission delay. For example, if a DAB system is used to shout in real time in a square, the speaker of the receiving terminal inevitably has a delay of at least 482ms, and this delayed sound will be re-transmitted back to the speaker's ear, resulting in a poor speaking experience. Many studies and experiments have demonstrated that in order for a real-time broadcast speaker to have a good speech experience, the delay of the broadcast system (from the speaker microphone to the receiving terminal loudspeaker) should be within 200ms, and the smaller the better.

In a commercial broadcast scenario at the city level, the speaker is located in the broadcast room and does not hear the broadcast sound of the outdoor receiving end device, which is not sensitive to such delay. However, with the expansion of the application scenarios of the DAB system, in the scenarios such as playground and hall, the speaker and the receiving end device are located at a close spatial position, and the speaker can hear the sound from the receiving end. When the voice is heard for 482ms before speaking, the human ears can feel delayed, poor interaction effect can be generated, and the speaking of the speaker is influenced. There is currently no good solution. Therefore, a new real-time voice transmission technology is urgently needed, and digital voice transmission with delay of no more than 200ms can be realized on the technology compatible with the existing DAB system. The difficulty is that the design must be carried out under the existing DAB standard framework, especially the physical layer of the DAB system cannot be changed, otherwise the existing chip and terminal cannot be used; the second is to delay no more than 200ms, and the smaller the better, to ensure good speaking experience.

In addition, in recent years, the DAB system is also widely used in emergency communication scenarios, and in emergency scenarios, real-time broadcasting with lower delay plays a crucial role. Therefore, the realization of the low-delay real-time voice broadcast can not only avoid the influence of the sensible delay on human ears, but also make the performance of the DAB system more advanced and the application scene more extensive.

Disclosure of Invention

In view of the above, the present invention provides a method and system for DAB low-delay real-time voice broadcasting.

In order to achieve the purpose, the invention provides the following technical scheme:

a method for DAB low-delay real-time voice broadcasting, comprising the steps of:

s1: collecting audio;

s2: coding by adopting EVS (enhanced Voice Services) with a code rate of 9.6kbps and a bandwidth of NB (Narrow Band ), WB (Wide Band, wideband) or SWB (Super Wide Band, ultra Wide Band) mode, and encapsulating each EVS frame into 1 DAB Fast Information Block (FIB);

s3: synthesizing DAB transmission frames by adopting a DAB mode III, and distributing continuous 6 FIBs containing EVS frames in 5 DAB transmission frames according to a certain rule for transmission;

s4: and receiving the DAB frame, decoding by adopting the EVS, and playing the audio.

Further, the corresponding encapsulation relationship between the EVS frame and the DAB transmission frame is: every 6 consecutive EVS frames are encapsulated into 5 consecutive DAB transmission frames.

Further, the code rate of the EVS is 9.6kbps, the period is 20ms, and the sampling rate is 32kHz, 16kHz or 8 kHz.

Further, the DAB frame is transmitted by adopting DAB mode III, and the transmission frame period is 24 ms.

A DAB low-delay real-time voice broadcasting system comprises an audio acquisition module, an enhanced voice service EVS coding module, a fast information channel FIC transmission module, a receiving module, an enhanced voice service EVS decoding module and a playing module;

the audio acquisition module is used for acquiring audio;

the EVS coding module for the enhanced voice service adopts EVS to code audio and then packages the audio into a Fast Information Block (FIB) of DAB;

the FIB is positioned in a fast information channel FIC of a DAB transmission frame and adopts a DAB mode III for transmission;

the DAB receiving module is used for receiving the DAB transmission frame and decoding an EVS frame from the DAB transmission frame;

the enhanced voice service EVS decoding module is used for decoding an EVS frame;

the playing module is used for playing the decoded audio.

Further, the transmission frame of the DAB consists of three channels, namely a synchronization channel, a fast information channel FIC and a main service channel MSC;

the Fast Information channel FIC includes 4 FIBs (Fast Information blocks); one FIB consists of a 30-byte data field containing useful data fields, end symbols and padding bits, and a 2-byte CRC (Cyclic Redundancy Check); the useful data field is 30 bytes at the maximum, and is composed of one or more FIG (Fast Information Group) fields, each of which contains a FIG header and a FIG data field, wherein the FIG header occupies 1 byte, and the maximum bearer data of one FIB is 29 bytes.

Further, the EVS has a frame period of 20ms, and is encapsulated in an FIB in an FIC, and the remaining 5-byte null is used for carrying control information.

Further, the corresponding encapsulation relationship between the EVS frame and the DAB transmission frame is: the first 4 EVS frames are encapsulated in turn in one FIB of the first 4 DAB frames, and the 5 th and 6 th EVS frames are encapsulated in two FIBs of the 5 th DAB frame.

The invention has the beneficial effects that: the DAB transmission mode adopts a mode III, the transmission frame period is 24ms, and compared with a transmission mode I, the delay caused by the transmission frame period is greatly shortened.

The encoding adopts EVS (enhanced Voice Services) encoding. The EVS is an ultra-wideband voice audio coding standard, is mainly used for real-time coding requirements, has lower delay compared with MP2 coding, and can well meet the requirements of real-time voice.

The collected audio is encoded by an EVS encoder and then encapsulated into DAB frames. Data is transmitted through a Fast Information Channel (FIC). The FIC does not need to undergo time interleaving, and 384ms delay caused by time interleaving can be avoided compared with transmitting data through the MSC.

The corresponding encapsulation relationship between the EVS frame and the DAB frame is as follows: every 6 consecutive EVS frames are packed into 5 consecutive DAB frames. In order to reduce the delay and reduce the encoding and decoding buffers to the maximum extent, only the necessary buffers of one frame EVS and one frame DAB are reserved so as to achieve the minimum delay.

Additional advantages, objects, and features of the invention will be set forth in part in the description which follows and in part will become apparent to those having ordinary skill in the art upon examination of the following or may be learned from practice of the invention. The objectives and other advantages of the invention may be realized and attained by the means of the instrumentalities and combinations particularly pointed out hereinafter.

Drawings

For the purposes of promoting a better understanding of the objects, aspects and advantages of the invention, reference will now be made to the following detailed description taken in conjunction with the accompanying drawings in which:

FIG. 1 is a diagram illustrating a conventional wireless real-time voice broadcast;

FIG. 2 is a schematic diagram illustrating delay of a conventional DAB voice broadcasting system;

FIG. 3 is a schematic structural diagram of a DAB low-delay real-time voice broadcasting system according to the present invention;

FIG. 4 is a structure diagram of a DAB transmission frame according to the present invention;

FIG. 5 is a block diagram of the FIB of the present invention;

FIG. 6 is a diagram of the packaging relationship of EVS and FIG according to the present invention;

fig. 7 is a diagram illustrating the encapsulation relationship between EVS frames and DAB transmission frames according to the present invention.

Detailed Description

The embodiments of the present invention are described below with reference to specific embodiments, and other advantages and effects of the present invention will be easily understood by those skilled in the art from the disclosure of the present specification. The invention is capable of other and different embodiments and of being practiced or of being carried out in various ways, and its several details are capable of modification in various respects, all without departing from the spirit and scope of the present invention. It should be noted that the drawings provided in the following embodiments are only for illustrating the basic idea of the present invention in a schematic way, and the features in the following embodiments and examples may be combined with each other without conflict.

Wherein the showings are for the purpose of illustrating the invention only and not for the purpose of limiting the same, and in which there is shown by way of illustration only and not in the drawings in which there is no intention to limit the invention thereto; to better illustrate the embodiments of the present invention, some parts of the drawings may be omitted, enlarged or reduced, and do not represent the size of an actual product; it will be understood by those skilled in the art that certain well-known structures in the drawings and descriptions thereof may be omitted.

The same or similar reference numerals in the drawings of the embodiments of the present invention correspond to the same or similar components; in the description of the present invention, it should be understood that if there is an orientation or positional relationship indicated by terms such as "upper", "lower", "left", "right", "front", "rear", etc., based on the orientation or positional relationship shown in the drawings, it is only for convenience of description and simplification of description, but it is not an indication or suggestion that the referred device or element must have a specific orientation, be constructed in a specific orientation, and be operated, and therefore, the terms describing the positional relationship in the drawings are only used for illustrative purposes, and are not to be construed as limiting the present invention, and the specific meaning of the terms may be understood by those skilled in the art according to specific situations.

Fig. 1 is a schematic diagram of a conventional wireless real-time voice broadcast, in which a speaker's voice is collected and transmitted, and then received by a receiver and played through a speaker. Because there is a delay between the speaker's voice and the speaker's voice, this delay can have a significant effect on the speaker as the speaker's voice travels to the speaker. The reason for the delay is shown in fig. 2, and mainly includes the following four aspects:

(1) a DAB transmission mode I is selected, the transmission frame period is 96ms, the period time is long, and the generation delay is high.

(2) Transmission through the MSC, which requires time interleaving, results in a delay of about 384 ms.

(3) The existing MP2/LC AAC audio format requires at least a codec buffer of 4 frames or more (including encoded 2 frames and decoded 2 frames), resulting in a large codec delay.

(4) DAB channel decoding will generate 2-24 ms delay.

Therefore, the delay in the conventional mode is at least 482 to 504 ms.

The structure of the DAB low-delay real-time voice broadcasting system of the present invention is shown in fig. 3.

In order to reduce the transmission delay as much as possible, DAB mode III transmission is adopted, the frame length is 24ms, and the real-time performance is higher compared with 96ms of mode I.

In order to solve the problem that MP2 and LC AAC audio coding and decoding adopted by DAB at present have larger delay, the audio coding adopts EVS coding. The EVS is an ultra-wideband voice audio coding standard, is mainly used for real-time coding requirements, and can be well suitable for real-time voice scenes. The sampling rate of the EVS is 8/16/32/48kHz, and the code rate of 5.9kbps-128kbps can be supported. The design adopts EVS (enhanced Voice Services) with code rate of 9.6kbps and bandwidth of NB (Narrow Band), WB (Wide Band) or SWB (Super Wide Band) mode to carry out coding, the sampling rate is 32kHz, 16kHz or 8kHz, and the frame period is 20 ms.

In order to solve the problem that the time interleaving exists when the MSC is used for transmitting the audio frame, so that 384ms delay is caused, the transmission is carried out in the FIC. As shown in fig. 4, the transmission frame of the DAB consists of three channels, which are a synchronization channel, a fast information channel and a main traffic channel. Compared with the MSC, the FIC does not need time interleaving, and 384ms time interleaving delay is avoided. Meanwhile, through FIC transmission, the delay of a decoding side is only 2ms, and compared with 2-24 ms of an MSC, the delay is lower.

In mode III, the FIC in one transmission frame contains 4 FIBs (Fast Information Block). The structure of the FIB is shown in fig. 5, and one FIB is composed of a 30-byte data field containing a useful data field, an end-pointer, and padding bits, and a 2-byte CRC (Cyclic Redundancy Check). The useful data field is 30 bytes at the maximum and is composed of one or more FIG (Fast Information Group) fields, each of which contains a FIG header and a FIG data field, wherein the FIG header occupies 1 byte, so that the maximum bearer data of one FIB is 29 bytes. The period of transmission mode III was 24ms and the maximum net data rate calculated per FIB was 9.66 kbps.

The EVS has a frame period of 20ms, when the code rate is 9.6kbps, 24 bytes can be transmitted per frame and encapsulated in one FIB in one FIC, and the remaining 5-byte null can be used for carrying control information. The package structure is shown in fig. 6.

The DAB frame period is 24ms, the EVS frame period is 20ms, and in order to ensure time synchronization, the corresponding relationship is known from the least common multiple relationship: every 6 consecutive EVS frames are packed into 5 consecutive DAB frames. To ensure the minimum delay of the whole packaging process, the optimal corresponding relationship is calculated as follows: the first 4 EVS frames are encapsulated in turn in one FIB of the first 4 DAB frames, and the 5 th and 6 th EVS frames are encapsulated in two FIBs of the 5 th DAB frame. The packaging relationship is shown in fig. 7.

The audio data is encoded by the EVS encoder before entering the DAB frame, and thus the start time of the EVS encoding needs to be at least one frame time earlier than the DAB encoding. In addition, the frame 5 encoding of DAB needs to be started after the frame 5 and frame 6 encoding of EVS is completed, and the minimum time difference between the two times is calculated to be 24 ms. Therefore, the lowest delay is achieved by starting the DAB frame 24ms after the EVS frame coding starts. The minimum delay produced by both codes is 48ms, the time from the start of coding the EVS first frame to the completion of coding the DAB first frame. Considering that the performance of the partial encoder environment is not sufficient to support 4ms accuracy, the time difference can be extended to the time of two EVS frames, i.e. 40ms, in this way, the total encoding delay is 64 ms. Therefore, the delay on the encoding side can be minimized to 48-64 ms by adopting the packaging relationship.

The EVS total delay is 32ms, including a one frame 20ms delay, a 0.94ms delay for input resampling at the encoding side and a forward delay of 8.75ms, a 2.31ms delay for time domain bandwidth extension at the decoding side. In combination with the above analysis, the total system delay of the design is 62-78 ms, and the minimum delay of 62ms includes an EVS encoding side delay of 24+0.94+ 8.75-33.69 ms, a DAB encoding side delay of 24ms, a channel decoding side delay of 2ms, and an EVS decoding side delay of 2.31 ms.

In summary, the total delay generated by EVS and DAB codec can be controlled within 100ms, compared with the delay of at least 420ms in the conventional real-time speech system, the design focuses on avoiding the time interleaving and greatly shortening the time consumed by codec, which is lower than the delay perceived by human ears, and realizing low-delay real-time speech broadcasting.

Finally, the above embodiments are only intended to illustrate the technical solutions of the present invention and not to limit the present invention, and although the present invention has been described in detail with reference to the preferred embodiments, it will be understood by those skilled in the art that modifications or equivalent substitutions may be made on the technical solutions of the present invention without departing from the spirit and scope of the technical solutions, and all of them should be covered by the claims of the present invention.

Claims

1. A DAB low-delay real-time voice broadcasting method is characterized in that: the method comprises the following steps:

s1: collecting audio;

s2: coding by adopting an EVS (enhanced Voice service) with a code rate of 9.6kbps and a bandwidth of NB (Narrow Band ), WB (Wide Band, wideband) or SWB (Super Wide Band, ultra Wide Band) mode, and packaging each EVS frame into a DAB Fast Information Block (FIB);

2. A method for DAB low delay real-time voice broadcasting as claimed in claim 1, characterized in that: the corresponding encapsulation relationship between the EVS frame and the DAB transmission frame is as follows: every 6 consecutive EVS frames are encapsulated into 5 consecutive DAB transmission frames.

3. A method for DAB low delay real-time voice broadcasting as claimed in claim 1, characterized in that: the EVS has a code rate of 9.6kbps, a frame length of 20ms and a sampling rate of 32kHz, 16kHz or 8 kHz.

4. A method for DAB low delay real-time voice broadcasting as claimed in claim 1, characterized in that: and transmitting the DAB frame by adopting a DAB mode III, wherein the transmission frame period is 24 ms.

5. A DAB low-delay real-time voice broadcasting system, characterized by: the system comprises an audio acquisition module, an enhanced voice service EVS coding module, a fast information channel FIC transmission module, a DAB receiving module, an enhanced voice service EVS decoding module and a playing module;

the audio acquisition module is used for acquiring audio;

the playing module is used for playing the decoded audio.

6. A DAB low-delay real-time voice broadcasting system according to claim 5, characterized in that: the DAB transmission frame consists of three channels, namely a synchronous channel, a Fast Information Channel (FIC) and a Main Service Channel (MSC);

7. A DAB low-delay real-time voice broadcasting system according to claim 5, characterized in that: the EVS one-frame is 24 bytes long and is encapsulated in the FIG of one FIB of 32 bytes in the FIC, and the remaining 5-byte vacant space of the FIG is used for carrying control information.

8. A DAB low-delay real-time voice broadcasting system according to claim 5, characterized in that: the corresponding encapsulation relationship between the EVS frame and the DAB transmission frame is as follows: every 6 consecutive EVS frames are encapsulated into 5 consecutive DAB transmission frames.

9. A DAB low-delay real-time voice broadcasting system according to claim 5, characterized in that: the corresponding encapsulation relationship between the EVS frame and the DAB transmission frame is as follows: the first 4 EVS frames are encapsulated in turn in one FIB of the first 4 DAB frames, and the 5 th and 6 th EVS frames are encapsulated in two FIBs of the 5 th DAB frame.

10. A DAB low-delay real-time voice broadcasting system according to claim 5, characterized in that: the EVS has a code rate of 9.6kbps, a frame length of 20ms and a sampling rate of 32kHz, 16kHz or 8 kHz.