ITU-R REPORT BT 2044-2004 Tolerable round-trip time delay for sound-programme and television broadcast programme inserts C Context and rationale《声音节目和电视广播节目插播的可容忍往返传播延迟时间的 背景和基本原理》.pdf
《ITU-R REPORT BT 2044-2004 Tolerable round-trip time delay for sound-programme and television broadcast programme inserts C Context and rationale《声音节目和电视广播节目插播的可容忍往返传播延迟时间的 背景和基本原理》.pdf》由会员分享,可在线阅读,更多相关《ITU-R REPORT BT 2044-2004 Tolerable round-trip time delay for sound-programme and television broadcast programme inserts C Context and rationale《声音节目和电视广播节目插播的可容忍往返传播延迟时间的 背景和基本原理》.pdf(14页珍藏版)》请在麦多课文档分享上搜索。
1、 Rep. ITU-R BT.2044 1 REPORT ITU-R BT.2044 Tolerable round-trip time delay for sound-programme and television broadcast programme inserts Context and rationale (Question ITU-R 35/6) (2004) CONTENTS Page 1 Scope 2 2 References and Bibliography 2 3 Terms and definitions . 3 4 Abbreviations and acronym
2、s 3 5 Causes of delay and echo . 4 5.1 Context 4 5.2 Historical background. 4 5.3 System design and configuration factors versus operating practices . 5 5.4 System design and configuration factors which cause delay and echo 6 5.4.1 Encode/decode processes. 6 5.4.2 Propagation delay. 6 5.5 Magnitude
3、of delay in various transmission media 7 5.6 Magnitude of delay in video codecs . 7 5.7 Magnitude of delay in audio codecs . 8 5.8 Circuit symmetry 10 5.9 Operating practices which cause echo 10 5.10 Sound to picture delay 10 6 Effects of delay and echo . 10 6.1 Audio delay and video delay 10 6.2 Va
4、riables affecting disturbance 11 6.3 Echoes and fusion . 11 6.4 Types of disturbance and their effects 12 6.5 Adaptation. 12 7 Summary. 12 2 Rep. ITU-R BT.2044 1 Scope This Report is intended to review the effects of delay and level of echo in an audio foldback loop in a broadcast production context
5、. It also reviews the effect of audio-video delay. It does not attempt to take into account the effect of additional reverberation and noise in the listening environment and assumes that there is no significant loss in signal quality from the monitoring system. 2 References and Bibliography 1 Techni
6、cal Report ETR 250 July 1996 Transmission and Multiplexing (TM); Speech commu-nication quality from mouth to ear for 3.1 kHz handset telephony across networks. European Telecommunications Standards Institute. http:/pda.etsi.org/pda/queryform.asp 2 Technical Report ETR 262 January 1996 Broadband inte
7、grated services digital network (B-ISDN); Asynchronous transfer mode (ATM); Video on demand (VOD) network aspects. European Telecommunications Standards Institute. 3 Technical Report ETR 275 April 1996 Transmission and Multiplexing (TM); Considerations on transmission delay and transmission delay va
8、lues for components on connections supporting speech communication over evolving digital networks. European Telecommunications Standards Institute. http:/pda.etsi.org/pda/queryform.asp 4 Recommendation ITU-R BT.1359 Relative timing of sound and vision for broadcasting. International Telecommunicatio
9、n Union (November 1998). 5 ETSI/GSM Recommendation 06.10 Version 3.2.0 February 1992 GSM Full Rate Speech Transcoding. European Telecommunications Standards Institute. http:/pda.etsi.org/pda/queryform.asp 6 DAVIS, P. 1995 A tutorial on MPEG audio compression. IEEE Multimedia, p. 60-74. 7 ONVURAL, R.
10、 1994 Asynchronous transfer mode networks: performance issues. Boston, Artech House. 8 Technical Report TR 100 815 V1.1.1 February 1999 Digital video broadcasting; guidelines for the handling of Asynchronous transfer mode signals in DVB systems. European Telecommuni-cations Standards Institute. 9 C.
11、S0014-0 Version 1.0 Enhanced Variable Rate Codec (EVRC) (CDMA2000 specification) Third Generation Partnership Project 2 (3GPP2) EIA/TIA (December 1999). 10 SIU-WAH WONG 1991 An Evaluation of 6.4 kbit/s Speech Codecs for Inmarsat-M System. IEEE, p. 629-632. 11 Recommendation ITU-R BT.1377 Labelling o
12、f video and audio apparatus throughput (processing) delay. International Telecommunication Union (1998). 12 EVEREST, F. A. 1994 The Master Handbook of Acoustics. TAB Books/McGraw-Hill. 13 CCITT Recommendation G.114 (1989) Mean one-way propagation time, Blue Book, Fasc. III.1. 14 CCITT Recommendation
13、 G.131 (1989) Stability and echo, Blue Book, Fasc. III.1. Rep. ITU-R BT.2044 3 15 ES 200 677 V1.2.1 Public Switched Telephone Network (PSTN) March 1998 Requirements for handset telephony. European Telecommunications Standards Institute. 16 LOCHNER, J. P. A. and BURGER, J. F. 1958 The subjective mask
14、ing of short time delayed echoes by their primary sounds and their contribution to the intelligibility of speech. Acustica, Vol. 8, 1, p. 1-10. 17 MEYER, E. and SCHODDER, G. R. 1952 ber den Einflu von Schallrckwrfen auf Richtungslokalisation und Lautstrke bei Sprache. Nachr. Akad. Wiss., Gttingen, 6
15、. 18 CHURCH, S. On beer recommended maximum decode delay: 3 ms. Satellite phone systems there are a number of competing systems including ICO, Iridium, AceS, AMSC-TMI and Inmarsat. These systems can be classed as low-Earth orbit (LEO) (ICO, Globalstar, Iridium and Teledesic),) or geostationary Earth
16、 orbit (GEO) (Inmarsat, Satphone, ASC, Thuraya, APMT, EAST). A typical GEO system, Inmarsat Mini-M, uses advanced multi-band excitation (AMBE) encoding at 4.8 kbit/s, with encode/decode delays comparable with those of GSM and CDMA terrestrial mobile telephone systems. In the GEO case however, the di
17、stance delay tends to make the encode/decode delays insignificant. Studio MPEG codec. Audio for broadcast may be embedded with video using MPEG-2 encoding, or it may be sent on separate ISDN/ATM lines. If embedded with video, the audio will be synchronized to the video. If sent separately, the audio
18、 may be uncompressed, giving a negligible encode/decode delay, or it may be compressed. Compression algorithms vary between manufacturers, so minimum and typical delays are cited. Due to the nature of audio signals a minimum bit rate for a given audio quality would deviate by a factor of 10 or more
19、over time. In general time-slots with a large demand on local bit rate are very short and surrounded by time-slots with very low bit rate. Most state of the art audio coding schemes therefore do some averaging of the bit rate over time to provide a constant bit rate. Depending whether the buffer for
20、 this averaging is in the encoder or decoder side, the encoding or the decoding delay is larger. There are several options for MPEG-1 audio compression: Layer I, Layer II and Layer III. Layer III is also known as MP3. MPEG-2 in addition to Layer I, II and III, offers advanced audio coding (AAC). In
21、MPEG-4 AAC was chosen as the baseline codec for natural audio and extended by several new tools and functionalities. In the context of round trip delay the low delay (LD) version of AAC coding is the most prominent extension. Rep. ITU-R BT.2044 9 Minimum delays for MPEG-1 Layer I, II, III, from 6, 1
22、8 End-to-end delay (including ISDN channel) for a commercial available audio codec 18. End-to-end delay of current audio coding schemes. MPEG audio encoded with MPEG video will generally have the same delay as the video. Layer Target bit rate (kbit/s) Compression ratio Theoretical minimum delay (ms)
23、 AAC-LD 64 12:1 20 Layer I 192 4:1 19 Layer II 128 6:1 35 Layer II 64 12:1 59 Coding Bit rate (kbit/s) Sampling rate (kHz) Typical delay (ms) AAC-LD stereo 128 48 60 AAC-LD mono 64 48 50 AAC stereo 128 48 172 Layer III stereo 128 48 326 Layer II stereo 128 48 224 Layer II stereo 128 24 (half mode) 3
24、98 G.722 64 48 10 Codec Bit rate (kbit/s) Algorithmic delay with bit reservoir set to zero (ms) 100% workload, burst transmission(ms) 100% workload, continuous transmission(ms) 30% workload, burst transmission (ms) 30% workload, continuous transmission(ms) Layer 2 192 34 Not available Not available
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