DRM数字中短波广播协议（ETSIES2019 V4.1.1）

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DRM（Digital Radio Mondiale，数字广播世界）是一项专门针对中短波广播领域的技术标准，由DRM组织制定，旨在提供高质量的数字音频广播解决方案，以替代传统的模拟中波（AM）和短波广播。ETSIE（European Telecommunications Standards Institute，欧洲电信标准化协会）发布的ETSI ES 2019 80 V4.1.1（2014年1月版本）是该技术的核心规范，它详细规定了DRM系统的体系结构、信号传输、编码方式以及兼容性要求。该规范主要关注以下几个关键知识点： 1. **广播技术**: DRM系统利用数字信号处理技术，能够实现高效、低噪声的音频传输，提高广播质量，支持多语言和多通道广播，满足全球各地听众的需求。 2. **信号标准**: ETSI ES 2019 80定义了DRM信号的标准格式，包括数据编码、调制方式、频谱利用率等，确保在全球不同地理位置和设备上都能稳定接收。 3. **参考文档**: RES/JTC-DRM-28是一个重要的参考资料，它可能包含技术细节、实施指南和最佳实践，有助于开发者和广播运营商遵循DRM标准进行实施。 4. **关键词**: "broadcasting"（广播）、"digital"（数字）、"DRM"（数字广播世界）和"radio"（无线电）强调了这个标准在广播行业的数字化转型中的核心地位。 5. **版权信息**: 用户需注意，尽管个人可以下载文档，但应从官方指定的网址获取，并确认PDF版本作为参考，以避免使用非官方或过时版本带来的潜在问题。此外，DRM标准可能会随着技术发展而更新或修订，因此用户应定期检查ETSIsite获取最新状态。 6. **联系信息**: ETSI总部位于法国索菲亚安提波利斯，用户可通过提供的电话和传真号码或网站获取更多支持和服务。 DRM数字中短波协议是现代广播行业中的一项重要技术，通过标准化的方式推动了音频广播的数字化升级，对于广播运营商、设备制造商以及内容创作者来说，理解和遵循这一规范是实现无缝数字广播服务的关键。

ETSI

ETSI ES 201 980 V4.1.1 (2014

01)

The source encoder and pre-coders ensure the adaptation of the input streams onto an appropriate digital transmission

format. For the case of audio source encoding, this functionality includes audio compression techniques as described in

clauses 4.3 and 5. The output of the source encoder(s) and the data stream pre-coder may comprise two parts requiring

different levels of protection within the subsequent channel encoder. All services have to use the same two levels of

protection.

The multiplexer combines the protection levels of all data and audio services as described in clause 6.

The energy dispersal provides a deterministic selective complementing of bits in order to reduce the possibility that

systematic patterns result in unwanted regularity in the transmitted signal.

The channel encoder adds redundant information as a means for quasi error-free transmission and defines the mapping

of the digital encoded information onto QAM cells as described in clause 7.

Cell interleaving spreads consecutive QAM cells onto a sequence of cells quasi-randomly separated in time and

frequency, in order to provide robust transmission in time-frequency dispersive channels. The pilot generator provides

means to derive channel state information in the receiver, allowing for a coherent demodulation of the signal.

The OFDM cell mapper collects the different classes of cells and places them on the time-frequency grid as specified in

clause 7.

The OFDM signal generator transforms each ensemble of cells with same time index to a time domain representation of

the signal. Consecutively, the OFDM symbol is obtained from this time domain representation by inserting a guard

interval as a cyclic repetition of a portion of the signal, as specified in clause 7.

The modulator converts the digital representation of the OFDM signal into the analogue signal in the air. This operation

involves digital-to-analogue conversion and filtering that have to comply with spectrum requirements as described in

annex E.

4.3 Audio source coding

Within the constraints of broadcasting regulations in broadcasting channels below 30 MHz and the parameters of the

coding and modulation scheme applied, the bit rate available for source coding is in the range from 8 kbit/s (half

channels) to ≈20 kbit/s (standard channels) to up to ≈72 kbit/s (double channels).

Within the constraints of broadcasting regulations in broadcasting channels above 30 MHz and the parameters of the

coding and modulation scheme applied, the bit rate available for source coding is in the range from 37 kbit/s to

186 kbit/s.

The system offers different audio source coding schemes:

• a subset of MPEG xHE-AAC (Extended High-Efficiency Advanced Audio Coding) for mono and stereo audio

broadcasting;

• a subset of MPEG-4 AAC (Advanced Audio Coding) including error robustness tools for mono and stereo

audio broadcasting, including:

- Spectral Band Replication (SBR), an audio coding enhancement tool that allows the full audio bandwidth

to be achieved at low bit rates;

- Parametric Stereo (PS), an audio coding enhancement tool relevant to SBR that allows for stereo coding

at low bit rates;

• MPEG Surround (MPS), an audio coding enhancement tool that allows for multichannel coding at low bit

rates.

For typical broadcast scenarios the use of xHE-AAC audio coding will achieve a better audio quality and therefore

should be preferred over AAC audio coding.

Provision is made for further enhancement of the audio system by linking two DRM signals together.

ETSI

ETSI ES 201 980 V4.1.1 (2014

01)

4.4 Transmission modes

4.4.1 Signal bandwidth related parameters

The current channel widths for radio broadcasting below 30 MHz are 9 kHz and 10 kHz. The DRM system is designed

to be used:

• within these nominal bandwidths, in order to satisfy the current planning situation;

• within half these bandwidths (4,5 kHz or 5 kHz) in order to allow for simulcast with analogue AM signals;

• within twice these bandwidths (18 kHz or 20 kHz) to provide for larger transmission capacity where and when

the planning constraints allow for such facility.

The current channel raster (where defined) for analogue radio broadcasting above 30 MHz is 100 kHz. The DRM

system is designed to be used with this raster.

These signal bandwidth related parameters are specified in clause 8.

4.4.2 Transmission efficiency related parameters

For any value of the signal bandwidth parameter, transmission efficiency related parameters are defined to allow a trade

off between capacity (useful bit rate) and ruggedness to noise, multipath and Doppler. These parameters are of two

types:

• coding rate and constellation parameters, defining which code rate and constellations are used to convey data;

• OFDM symbol parameters, defining the structure of the OFDM symbols to be used as a function of the

propagation conditions.

4.4.2.1 Coding rates and constellations

As a function of the desired protection associated within each service or part of a service, the system provides a range of

options to achieve one or two levels of protection at a time. Depending on service requirements, these levels of

protection may be determined by either the code rate of the channel encoder (e.g. 0,6, etc.), by the constellation order

(e.g. 4-QAM, 16-QAM, 64-QAM), or by hierarchical modulation. Detailed definition of these options is given in

clause 7.

4.4.2.2 OFDM parameter set

The OFDM parameter set is presented in this clause. The specification of the signal waveform is given in clause 8.

These values are defined for different propagation-related transmission conditions to provide various robustness modes

for the signal. In a given bandwidth, the different robustness modes provide different available data rates. Table 1

illustrates typical uses of the robustness modes.

Table 1: Robustness mode uses

Robustness mode

Typical propagation conditions

A Gaussian channels, with minor fading

B Time and frequency selective channels, with longer delay spread

C As robustness mode B, but with higher Doppler spread

D As robustness mode B, but with severe delay and Doppler spread

E Time and frequency selective channels

The transmitted signal comprises a succession of OFDM symbols, each symbol being made of a guard interval followed

by the so-called useful part of the symbol. Each symbol is the sum of

sine wave portions equally spaced in

frequency. Each sine wave portion, called a "cell", is transmitted with a given amplitude and phase and corresponds to a

carrier position. Each carrier is referenced by the index

k, k belonging to the interval

[]

maxmin

,kk ( 0=k corresponds to

the reference frequency of the transmitted signal).

ETSI

ETSI ES 201 980 V4.

1.1 (2014

01)

5 Source coding modes

5.1 Overview

The audio source coding options in the DRM system are shown in figure 2. As described in clause 4.3, the DRM system

offers two audio codecs, xHE-AAC and AAC (in combination with the SBR and PS tools). Optionally, MPS can be

used to enable multichannel coding. The encoded audio is composed into audio super frames of constant length.

Multiplexing and UEP of audio services is done by means of the multiplex and channel coding units. Audio specific

configuration information is transmitted in the SDC (see clause 6.4.3.10).

DRM Audio Source Encoding

MPEG Surround

Encoder

(config. depnd.)

MPEG PS

Encoder

(config. depnd.)

audio

signal

SBR Encoder

(configuration

dependent)

xHE-AAC Encoder

(stereo coding & SBR handled internally/automatically)

AAC

Encoder

Audio

super framing

mux &

channel

coding

bitstream

from mux

& channel

decoding

AAC

Decoder

xHE-AAC

Decoder

Audio

super frame

demux

SBR / PS

Decoder

MPEG Surround

Decoder

DRM Audio Source Decoding

multichannel

audio output

mono/stereo

audio output

binaural stereo

output for

headphones

Figure 2: Audio source coding overview

5.1.1 Extended HE-AAC audio coding (xHE-AAC)

For generic coding of both audio and speech content at all bit rates, a subset of the MPEG xHE-AAC toolbox chosen to

best suit the DRM system environment is used. For example a standard configuration for use in one short wave channel

could be 16 kbit/s stereo.

Specific features of the xHE-AAC stream within the DRM system are:

• Bit rate: xHE-AAC can be used at any bit rate. The granularity of the xHE-AAC bit rate is 20 bit/s for

robustness modes A, B, C and D and 80 bit/s for robustness mode E.

• Sampling rates: Permitted sampling rates for the use of xHE-AAC within DRM are selected such that the

interface of the xHE-AAC codec towards the surrounding application can easily accept or provide 48 kHz

audio signals, respectively. The actual core sampling rate is selected by the encoder upon initialization to

ensure the best possible audio signal quality and is typically not visible to higher layers of processing.

• Audio super framing: To ensure the best possible audio quality particularly at lower bit rates, the xHE-AAC

encoder can flexibly assign the available bit rate within certain constraints to each audio frame. Audio super

frames - as generated by the xHE-AAC audio encoder and inserted into the DRM logical frames - always have

a constant size. However, the number of audio frames per audio super frame is not fixed, and audio frames

may span audio super frames. This flexibility is achieved by a slight adjustment to the audio super frame

header configuration used for the AAC codec in DRM. One audio super frame is always placed in one DRM

logical frame in robustness modes A, B, C and D and in two logical frames in robustness mode E (see clause

6). In this way no additional synchronization is needed for the audio coding. Retrieval of frame boundaries is

also taken care of within the audio super frame.

ETSI

ETSI ES 201 980 V4.1.1 (2014

01)

• UEP shall not be used with xHE-AAC.

5.1.2 AAC audio coding

For generic audio coding, a subset of the MPEG-4 Advanced Audio Coding (AAC) toolbox chosen to best suit the

DRM system environment is used. For example a standard configuration for use in one short wave channel could be

20 kbit/s mono.

Specific features of the AAC stream within the DRM system are:

• Bit rate: AAC can be used at any bit rate. The granularity of the AAC bit rate is 20 bit/s for robustness modes

A, B, C and D and 80 bit/s for robustness mode E.

• Sampling rates: permitted sampling rates are 12 kHz and 24 kHz for robustness modes A, B, C and D and

24 kHz and 48 kHz for robustness mode E. 48 kHz is only permitted if the SBR tool is not used.

• Transform length: the transform length is 960 to ensure that one audio frame corresponds to 80 ms or 40 ms

(robustness modes A, B, C and D) or to 40 ms or 20 ms (robustness mode E) in time. This is required to allow

the combination of an integer number of audio frames to build an audio super frame of 400 ms (robustness

modes A, B, C and D) or 200 ms (robustness mode E) duration.

• Error robustness: a subset of MPEG-4 tools is used to improve the AAC bit stream error robustness in error

prone channels (the MPEG-4 EP tool is not used).

• Audio super framing: 5 or 10 audio frames are composed into one audio super frame. For robustness modes A,

B, C and D, the respective sampling rates are 12 kHz and 24 kHz producing an audio super frame of 400 ms

duration; for robustness mode E, the respective sampling rates are 24 kHz and 48 kHz producing an audio

super frame of 200 ms duration. The audio frames in the audio super frames are encoded together such that

each audio super frame is of constant length, i.e. that bit exchange between audio frames is only possible

within an audio super frame. One audio super frame is always placed in one logical frame in robustness modes

A, B, C and D and in two logical frames in robustness mode E (see clause 6). In this way no additional

synchronization is needed for the audio coding. Retrieval of frame boundaries and provisions for UEP are also

taken care of within the audio super frame.

• UEP: better graceful degradation and better operation at higher BERs is achieved by applying UEP to the

AAC bit stream. Unequal error protection is realized via the multiplex/coding units. For robustness mode E,

the length of the higher protected part of an audio super frame shall be a multiple of 2 bytes.

SBR coding

To maintain a reasonable perceived audio quality at low bit rates, classical audio or speech source coding algorithms

need to limit the audio bandwidth and to operate at low sampling rates. It is desirable to be able to offer high audio

bandwidth also in very low bit rate environments. This can be realized by the use of Spectral Band Replication (SBR).

The purpose of SBR is to recreate the missing high frequency band of the audio signal that could not be coded by the

encoder. In order to do this in the best possible way, some side information needs to be transmitted in the audio

bitstream, removing a small percentage of the available data rate from the audio coder. This side information is

computed on the full bandwidth signal, prior to encoding and aids the reconstruction of the high frequencies after

audio/speech decoding.

SBR exists in two versions. The version difference is only reflected in the decoder design. High Quality SBR uses a

complex filterbank whereas Low Power SBR uses a real-valued filterbank plus anti-aliasing modules. The Low Power

version of SBR offers a significant reduction in complexity as compared to the High Quality version without

compromising too much on audio quality. AAC + SBR is defined in MPEG-4 Audio (High Efficiency AAC profile) [2].

PS coding

For improved performance at low bit rate stereo coding, a Parametric Stereo (PS) coder is available. The PS tool can be

used when running the configuration AAC + SBR (MPEG High Efficiency AAC profile). The general idea with PS

coding is to send stereo image describing data as side information along with a downmixed mono signal. This stereo

side information is very concise and only requires a small fraction of the total bit rate allowing the mono signal to have

maximum quality for the total bit rate given.

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