DRM数字调幅广播标准：ETSI_ES_201_980_v2.3.1详解

需积分: 11 187 浏览量更新于2024-07-31 1 收藏 1.35MB PDF 举报

"数字调幅广播标准ETSI_ES_201_980_v2.3.1" 是一个由欧洲电信标准协会（ETSI）制定的规范，用于定义Digital Radio Mondiale (DRM) 数字调幅广播系统的技术规格。这个标准在2008年2月更新至版本2.3.1，旨在提升传统调幅广播的质量和效率。 DRM（Digital Radio Mondiale）是一种全球性的数字广播标准，特别针对中短波频率进行设计，旨在替代传统的模拟调幅广播。该技术不仅提高了音频质量，还提供了数据服务、文本信息和多语言广播等附加功能。DRM系统通过高效的编码和压缩技术，能够在有限的频谱资源中传输更多的信息。 ETSI ES 201 980 v2.3.1标准详细描述了DRM系统的各个方面，包括： 1. **系统架构**：DRM系统由发射端、传输链路和接收端组成，涵盖了信号的编码、调制、解调以及相关的网络和接口规范。 2. **编码与压缩**：为了适应不同带宽和传输条件，DRM使用了高效的音频编码算法，如AAC+ (Advanced Audio Coding Plus)，它能在较低的数据速率下提供高质量的音频体验。 3. **调制方式**：DRM支持多种数字调制方式，如QAM（Quadrature Amplitude Modulation）和COFDM（Coded Orthogonal Frequency Division Multiplexing），以适应不同的频段和传播环境。 4. **错误纠正**：标准中包含了错误检测和纠正机制，如前向纠错编码（FEC），以确保在信道条件不佳时仍能可靠传输数据。 5. **服务质量**：DRM系统可以动态调整编码参数，以应对不断变化的无线电传播条件，确保稳定的服务质量。 6. **元数据与交互**：DRM广播还可以携带元数据，如节目信息、艺术家和歌曲名称，甚至支持听众与广播台的互动功能。 7. **兼容性**：考虑到全球范围内的广播网络，DRM标准必须与现有的模拟广播系统兼容，并且能够平滑过渡到数字广播。 8. **安全与版权保护**：标准可能还包括对内容加密和版权保护的机制，以防止非法复制和分发。 9. **测试与验证**：为了确保设备间的互操作性和符合标准，ETSI还会定义一系列的测试程序和合规性要求。 "数字调幅广播标准ETSI_ES_201_980_v2.3.1"是DRM技术的核心规范，它定义了从信号生成到接收的整个过程，确保了数字广播的高效、可靠和高质量。该标准对于推动全球广播业的数字化进程至关重要。

ETSI

ETSI ES 201 980 V2.3.1 (2008

02)

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 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).

In order to offer optimum quality at a given bit rate, the system offers different source coding schemes:

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

stereo audio broadcasting;

• a subset of MPEG-4 CELP speech coder for error robust speech broadcasting in mono, for cases when only a

low bit rate is available or especially high error robustness is required;

• a subset of MPEG-4 HVXC speech coding for very low bit rate and error robust speech broadcasting in mono,

especially well suited also for speech data base applications;

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

achieved at low bit rates. It can be applied to AAC, CELP and HVXC;

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

bit rates.

The bit-stream transport format of the source coding schemes has been modified to meet the requirements of the DRM

system (audio superframing). Unequal Error Protection (UEP) can be applied to improve the system behaviour in error

prone channels.

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

ETSI

ETSI ES 201 980 V2.3.1 (2008

02)

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.

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

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 V2.3.1 (2008

02)

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

time. This is required to harmonize CELP and AAC frame lengths and thus to allow the combination of an

integer number of audio frames to build an audio super frame of 400 ms duration.

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

prone channels.

• Audio super framing: 5 (12 kHz) or 10 (24 kHz) audio frames are composed into one audio super frame, which

always corresponds to 400 ms in time. 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 (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.

5.1.2 MPEG CELP coding

MPEG CELP speech coding is offered in the DRM system to allow for reasonable speech quality at bit rates

significantly below the standard rate (for example "half rate" operation at 8 kbit/s). Possible scenarios for the use of the

speech coder are:

• Dual/triple speech applications: instead of one audio program at 20 kbit/s to 24 kbit/s, the channel contains two

or three speech signals of 8 kbit/s to 10 kbit/s each, allowing simultaneous speech transmissions.

• Speech services in addition to an audio service.

• Simulcast transmissions: in case of analogue/digital simulcast only bit rates as low as 8 kbit/s may be

available.

• Very robust speech applications: due to its nature a speech coder can be expected to offer higher robustness

against channel errors. Therefore 8 kbit/s speech coding can be used to do ultra robust speech coding in one

channel.

Basic features of MPEG CELP coding are:

• 8 kHz or 16 kHz sampling rate;

• Bit rates between 4 kbit/s and 20 kbit/s;

• error robustness;

• composition of an integer number of CELP frames to build one audio super frame.

5.1.3 MPEG HVXC coding

MPEG-4 HVXC (Harmonic Vector eXcitation Coding) speech coding is offered in the DRM system to allow for

reasonable speech quality at very low bit rates such as 2,0 kbit/s. The operating bit rates of HVXC open up new

applications for DRM such as:

• Speech services in addition to an audio service.

• Multi-language application.

• Solid-state storage of multiple programs such as news, data base in a card radio (e.g. total of about 4,5 hours of

radio programs can be stored in 4 MByte Flash memory).

• Time scale modification for fast playback/browsing of stored program.

• Highly error robust transmission with or without hierarchical modulation scheme.

剩余194页未读，继续阅读

qihongshao

粉丝: 0
资源: 7

DRM数字调幅广播标准：ETSI_ES_201_980_v2.3.1详解

ETSI EN 301 192 V1.4.1：2004年数字视频广播数据传输标准

DMB标准ETSI TS 102428 V1.2.1：数字广播与视频服务技术规格

欧洲标准 ETSI EN 300 401：数字音频广播（DAB）技术

ETSI_ETS_300_706.rar_300_EN 300 706_ETS 3_ETSI_etsi 300 396-7

ETSI.rar_ETSI TETRA_TETRA ETSI_tetra_wimax

etsi_ts_102_693_v1.1.2.pdf_DABETSI102693_

ETSI_TS_102_006_v1.3.1_(2004-05)

etsi_ts_102600_102699_102658

E-Sim卡 ETSI_TS_102671V12.0.1使用手册

etsi_300_799_June_1997.pdf_300_DABETSI300799_源码

最新资源