Audio in Next Generation DVB Broadcast Systems

Auteurs : Roland Vlaicu
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Audio in Next Generation DVB  Broadcast Systems


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	    <date dateType="Created">Mon 4 Sep 2017</date>
	    <date dateType="Updated">Mon 4 Sep 2017</date>
            <date dateType="Submitted">Sat 17 Feb 2018</date>
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D LA TECHNOLOGIE DE LA HAUTE DÉFINITION Audio in Next Generation DVB Broadcast Systems Par RolandVLAICU Dolby Laboratories Mots clés Audio, HauteDéfinition, Télévision, Télédiffusion, Son«surround» The extensive new requirements for audio delivery in next generation broadcast systems such as High Definition Television are considered, and a resulting new candidate audio standard is introduced. Introduction This document discusses the requirements for an audio coding scheme to address both near and long term requirements of DVB broadcasting such as High Definition Television (HDTV). A number of factors have been considered in determining the characteristics of a suitable audio coding scheme : . Requirements for new DVB services such as HDTV and services using next generation video codees such as H.264 . Opportunities for improving the audio performan- ce of current DVB broadcast services . Impact of a new audio coding scheme on broadcast production practices . Impact of a new audio coding scheme on consumer hardware and listening environments The scope of this document concerns MPEG transport stream based DVB services as defined in ETSI standards document TS 101 154 [l]. Requirements for existing DVB services, near and long term The audio codecs previsously specified within TS 101 154 offer solutions for many of the requirements facing DVB broadcasters - requirements for high quality audio at low bit rates, multichannel audio services, guaranteed connectivity with consumer hardware through existing IEC 61937 interfaces (S/PDIF or ToslinkTM), and control by a broadcaster over the consumer listening experience through the use of audio metadata. Each codec previously specified in TS 101 154 meets some of these require- ments, but there is currently no single codec that meets all of them. For example, the AC-3 (Dolby Digital) codec can deliver up to 5.1 channels of audio, offers broadcasters full control over the listening experience for all consumer environments through comprehensive metadata control, and offers standardised connectivity via IEC 61937 to over 40 millions existing consumer A/V systems. However, the AC-3 codec is not optimised for low bit SSENTIEL SYNOPSIS Leschaînesde télévision ont accruleur niveaud'exigencede qua- lité pour le déploiement de nouveauxsystèmes de téiédiffusion tels que la TélévisionHaute Définition. Ces exigencesincluent la capacitéde diffuser des flux audioaussi bien en mono qu'en 5.1 et au-delàavec une efficacité supérieureaux systèmes existants, tout en maintenantla compatibilitéaveclesappareilshome-cinéma actuels. Un nouveausystème de compressionde l'audio, appelé « EnhancedAC-3 »a été développépour satisfaireces exigences et est actuellementsoumis pournormalisationauprèsdesconsor- tiums DVBet ATSC. Broadcastershavesignificant new requirementsfor audio delive- ry in next generationbroadcastsystems such as High-Definition Television. These include the capability to deliver soundtracks from mono to 5.1-channelsand beyond with greater efficiency than with current systems, but alsoto maintaincompatibilitywith existing consumer home cinema systems. A new audio delivery system, referred to as EnhancedAC-3, has been developed to meet these requirements, and is currently being considered by DVBandATSCfor standardisation. REE HORSSÉRIENT Septembre2005 Audio in Next Generation DVB Broadcast Systems rate performance. In contrast, the MPEG-4 HE-AAC codec delivers excellent performance at low bit rates, but does not offer IEC 61937 connectivity to consumers' existing A/V systems for delivery of 5.1 channel content, and does not offer mandated implementation and inter- operability testing of metadata in encoder and decoder products. When considering the feature set used by current DVB services, a new audio codec should offer at least the following : . Support for mono to 5.1 channel capability . Comprehensive metadata support, mandated in both encoder and decoder - ail parameters under encoder control : - Dialogue normalization to ensure consistent liste- ning levels between programmes - Downmix to ensure backward compatibility with matrix surround, stereo and mono systems - Control of dynamic range to ensure optimal reproduction for all consumer listening environments . Delivery of discrete 5.1 channel audio to current install base ofA/V receivers via IEC 61937 inter- faces, and support for other emerging digital inter- face standards . Improved bitrate efficiency compared with audio codecs currently in use in DVB services, in line with efficiency gains of new video codecs . Licensing costs and terms in line with existing audio codecs . Encoder and decoder products subject to interope- rability testing to ensure consistent performance In addition to these requirements for core broadcast services, there is also opportunity to improve the current provisions for deployment of audio description services for visually and hearing impaired. While relative levels between audio description (AD) and main programme services can be controlled both by the broadcaster and the listener, variations in loudness and dynamic range bet- ween programmes leads to a need for regular adjustments to listening levels by the consumer. A new audio codec should meet the following requi-CD rements to deliver improved AD services : . Metadata control of dialogue levels to ensure a consistent relative level between Main and AD programmes. . Metadata control of the dynamic range of the main programme to ensure that AD services are clearly audible at ail times. . Metadata to control mixing of main programme and c Zn AD services in a broadcast receiver should be suppor- ted, to remove the need for frequent manual adjust- ment of levels in the broadcast receiver. Support for mixing AD services with multichannel as well as ste- reo programme content should be available To simplify implementation in a broadcast recei- ver, the ability to deliver both main programme and AD services as a single stream that can be decoded and mixed using a single decoder in the broadcastreceiveris desirable Requirements for new services Standards and technologies are being developed for the next generation of DVB broadcast services, a new audio codec must be flexible enough to adapt to these new requirements. Applications such as high definition television and interactive services present new opportu- nities for audio services. A new audio codec should satis- fy at least the following requirements to meet the demands of future broadcast services : . Be able to deliver audio quality improvements to match video quality improvements of HD broadcasting . Flexibility to deliver more than 5.1 channels of audio to match future feature film mixing formats . Support for mixing of interactive audio content with main programme audio, including multichan- nel content. . Deployment of multiple programmes in a single stream, enabling multiple languages, director's commentaries or other advanced services, all controlled by mixing metadata, to be decoded using a single decoder in the broadcast receiver Impact on broadcast production environment The adoption of a new audio codec for final broadcast should have minimal affect on a broadcaster's working methods, and should certainly not adversely affect them. In the case of stereo audio services, the process of crea- ting audio content does not differ greatly from codec to codec - the selection of encoding settings need be done only once, based upon the target quality of a broadcas- ter's service and the capability of the codec selected. If the selected codec supports control of programme loud- ness and dynamics through metadata, this will also need to be factored into the production process. When considering multichannel audio services, the inter- facing of a new emission codec with a broadcast production environment must be carefully considered. Multichannel audio presents a number of challenges to a broadcaster ; dis- tribution of content within a broadcast facility equipped for stereo only content, and the task of creating audio metadata to ensure that 5.1 programming can deliver optimal back- ward compatibility with all listening environments. When considering the integration of a new audio codec with a broadcast production environment it is important that the level of metadata functionality should at least REE HORSSÈRtENl Septembre2005 D LA TECHNOLOGIE DE LA HAUTE DÉFINITION match, and preferably exceed that of current codecs, to ensure that a broadcaster's ability to deliver consistent audio quality is maintained, while allowing flexibility for development of future services. Simple interfacing bet- ween multichannel production equipment and transmis- sion encoders for both audio and metadata is desirable. Impact on consumer products and listening environments A new audio codec should offer performance impro- vements for the consumer, while ensuring simple integration into the current consumer listening environment, and offering flexibility for future developments in consumer product design and functionality. To ensure a consistent listening experience for all consumers, a new audio codec should meet the following requirements : . Decoders must maintain compatibility with exis- ting consumer AN receivers and IEC 61937 inter- faces when delivering discrete 5.1 channel content, without introducing excessive complexity to the decoder design . To remove the need for audio simulcasting, ail decoders must be able to receive and decode, as required, multichannel audio services, in order to deliver either a matrix surround, stereo or mono downmix as required. (The new audio codec should not remove the possibility of audio simul- casting if desired by a broadcaster) . Decoder complexity should be in line with current designs for an equivalent feature set . Metadata created during the production process must be supported by all decoders. (If this is not the case then the benefits of using metadata are usually lost) . The new codec should be compatible with emerging and future digital interface standards without intro- ducing excessive complexity to the decoder design . Licensing costs and terms in line with existing audio codecs A new solution Considering these requirements, a new audio coding scheme has been developed for use in next generation appli- cations. This coding scheme, referred to as Enhanced AC-3 (E-AC-3) or Dolby Digital Plus, has already been awarded Candidate Standard status within ATSC and has been selec- ted as mandatory technology for HD-DVD players. The scheme is also currently being considered by DVB for next generation broadcast services, and is expected to be standar- dised by ETSI shortly as TS 102 366 Annex E. Applications for which E-AC-3 is well suited include lower data-rate carriage of audio and its conversion to the AC-3 coding standard for playback on today's installed base of audio/video entertainment equipment ; interactive multimedia capabilities that allow the combination of streamed content with a main audio program ; the repro- duction of greater than 5.1 channels for the support of playback of existing and future cinema content ; and the efficient transcoding of AC-3 program content to lower data-rate E-AC-3 bitstreams and conversion back to AC- 3 for playback on the installed base of AC-3 decoders. Compatible Lower Data-Rate Carriage There are a growing number of applications that require lower data rates, but also require compatibility to the existing broadcast-reception and audio/video deco- ding infrastructure. The E-AC-3 system is an excellent solution for these applications because of its inherent lower tandem coding losses with AC-3 and its greater coding efficiency provi- ded by new coding tools. Because of the very large ins- talled base of AC-3 decoders, the E-AC-3 system has been designed to permit a very low loss conversion to standard AC-3 over a digital audio interconnect such as S/PDIF and decoding by a standard AC-3 decoder. The conversion stage is a special form of transcoder that minimizes quality degradations resulting from tan- dem coding losses. This is feasible with the use of the same filterbank, transform block alignment, bit-alloca- tion process, and basic framing structure as conventional AC-3. A Next Generation Television Set Top Box Application The next generation television application is very similar to the conventional AC-3 reception paradigm except that the need for greater channel capacity requires the transmission of audio programming at lower data rates than is typical for AC-3 applications. Traditionally, E-AC-3/AC-3 1 - 5l ch AC-3 Converter/Decoder 32-448kbps S/PDIF A H ne R ceiverlom c C1 - 5.1 ch E-AC-3 E-AC-3/AC-31-5.1ch EAC-3 demder ps to AC-3 640kbps Converter & Decoder 2 ch PCM Figure 1. Television set-top box converterldecoder (existing AC-3 devices in grey) [2]. REE HORSSÉRIE WI Septembre2005 Audio in Next Generation DVB Broadcast Systems AC-3 bas been employed at 128 - 192 kbps for stereo and 384 - 448 kbps for 5. 1 -channel applications. The use of the new coding tools in E-AC-3 allows for the practical use of lower data rates while permitting efficient conver- sion into a conventional AC-3 bitstream at 640 kbps for compatibility with existing home theaters. Figure 1 shows the configuration of this converter/decoder. The device shown in figure 1 accepts both the E-AC-3 and AC-3 bitstreams but always outputs standard AC-3 as the consumer switches channels within the network. The greater efficiency of the E-AC-3 system allows for a greater number of programs within the broadcast system while preserving full functionality for legacy receiver hardware, shown with a gray interior. New Coding Tools for Greater Efficiency Coding efficiency has been increased to allow for the beneficial use of lower data rates. This is accomplished using an improved filterbank, improved quantization, enhanced channel coupling, spectral extension, and a technique called transient prc-noise processing, designed to reduce excess noise before transients. The improved filterbank is a result of the addition of a second stage Discrete Cosine Transform (DCT) after the existing AC-3 filterbank, when audio with stationary characteristics is present. This converts the six 256-coef- ficient transform blocks into a single 1536-coefficient hybrid transform block with increased frequency resolu- tion. This increased frequency resolution is combined with 6-dimensional Vector Quantization (VQ) and Gain Adaptive Quantization (GAQ) to improve the coding efficiency for " hard to code " signals such as pitch pipe and harpsichord. VQ is used to efficiently code frequen- cy bands requiring lower accuracies, while GAQ pro- vides greater efficiency when higher accuracy quantiza- tion is required. Improved coding efficiency is also obtained through the use of channel coupling with phase preservation. This method expands on the AC-3 method of employing a high frequency mono composite channel which reconsti- tutes the high-frequency portion of each channel on deco- ding. The addition of phase information and encoder- controlled processing of spectral amplitude information sent in the bitstream improves the fidelity of this process so that the mono composite channel can be extended to lower frequencies than was previously possible. This decreases the effective bandwidth encoded, and thus increases the coding efficiency. Another powerful tool added is spectral extension. This method builds on the channel coupling concept by replacing the upper frequency transform coefficients with lower frequency spectral segments translated up in frequency. The spectral characteristics of the translated segments are matched to the original through spectral modulation of the transform coefficients. An additional technique to improve audio quality at low data rates is the use of transient pre-noise processing. This is a post-decoding process that reduces pre-noise error via time scaling synthesis techniques which reduce pre-noise duration and therefore audibility of transient artifacts common to bit-rate-reduction audio coders. The parameters used in the post-decoding time scaling synthesis processing, which employs auditory scene analysis, are provided by metadata calculated by the encoder and transmitted in the E-AC-3 bitstream. Conclusion 5.1 surround sound is an important component of the HDTV experience. Existing MPEG2-based HDTV ser- vices in the US and Australia already offer 5.1 surround sound using the AC-3 technology as documented by DVB and ATSC. With the move to next generation video coding systems, broadcasters have many new require- ments for audio delivery. The E-AC-3 system has been developed to meet these requirements whilst also main- taining compatibiiity with over 40 millions existing consumer home cinema systems. References [11 Standards document ETSI TS 101 154, version 1.6.1, "DVB : Implementation Guidelines for the use of Videoand Audio Coding in Broadcasting Applications ",January 2005. [2] Louis D. Fielder & al, " Introduction ta Dolby Digital Plus,an Enhancement to the Dolby Digital Coding System ", 117th AES Convention, October 2004. a u e u Roland Vlaicugraduatedfrom the Technical Universityof Munich,Germanyln 1997with a Master'sDegreein specialising ln MPEGA Video compression. During his studieshespentasemesterlntheUKworkingwiththe Universityof CambridgeonanInternshipproject.Healsoworked part-time since1990at ProSieben Télévision, Germany'ssecond largestcommercialbroadcasterwhichhethenjoinedfull-time in 1998to workasa systemsengineerin thedepartmentfordistr ! - butionnetworks. Afterhavingworkedasa projectmanageronthe introductionof DolbyDigital5.1 soundto ProSieben'smainservicehe joined Dolby Laboratories in July2000asa BroadcastConsultant.His currenttasks include working withbroadcasterslnEuropeand ASlaonthe implementationof Dolby Digitalonnewandexisting DigitalTVservices. REE HORS SÉRIE WI Septembre2005