- 您現(xiàn)在的位置：買賣IC網(wǎng) > PDF目錄382907 > CONFERENCEPAPERREPRINT Conference Paper Reprint - Multimedia over FDDI (Paper) PDF資料下載
參數(shù)資料
| 型號： | CONFERENCEPAPERREPRINT |
| 英文描述： | Conference Paper Reprint - Multimedia over FDDI (Paper) |
| 中文描述： | 會(huì)議文件再版-在FDDI（多媒體文件） |
| 文件頁數(shù)： | 4/28頁 |
| 文件大?。?/td> | 120K |
| 代理商： | CONFERENCEPAPERREPRINT |
第1頁第2頁第3頁當(dāng)前第4頁第5頁第6頁第7頁第8頁第9頁第10頁第11頁第12頁第13頁第14頁第15頁第16頁第17頁第18頁第19頁第20頁第21頁第22頁第23頁第24頁第25頁第26頁第27頁第28頁
network is heavily loaded, a station applying a large load
will encounter significant delays before transmission.
Thus, although Ethernet under light load offers excellent
average latencies, at high load it offers little or no bound
on the network access time.
Public networks or the telecom networks, on the other
hand, are typically optimized for circuit-switching
applications such as voice which requires low bandwidth
and low latency. These networks typically cannot provide
low access delay to bursty data.
In order to evaluate the feasibility of using existing
LANs in multimedia applications, we examine the issues
in sound and video transmissions such as bandwidth
requirements, latency, jitter and maximum number of
sessions. The characteristics of sound and video and the
requirements of multimedia on networks are examined.
Finally, the feasibility of multimedia over FDDI is
explored.
3: Definitions
Available bandwidth
is the bandwidth which is actually
available for valid transmissions. Available bandwidth
can also be measured in terms of network efficiency.
Thus, in FDDI efficiency h = (T - D) / T; where T= target
token rotation time in ms and D = ring latency in ms.
If D=0.1 ms and T = 10 ms, then h = 99% and available
bandwidth is 99 Mbps whereas total bandwidth is 100
Mbps.
Latency
is the average end to end message delay which
includes time for A/D conversion (if any), sample and
encode, packetization, queuing delay, transmission delay,
propagation delay, receive delay, decode, and
presentation.
Jitter
is the maximum instantaneous variation in object
presentation time. If the object is a packet, then the
maximum inter-packet arrival time variation is defined as
packet jitter.
Session
is defined as an interactive communications
dialogue between two or more users. Thus a telephone
conversation between two people is a session which
consumes a portion of the available bandwidth.
4: Characteristics of Sound
We classified sound as human speech and music.
Human speech or voice is typically in the 0-4 khz
spectrum. The bandwidth of music discernible by the
human ear is 20-25 khz (high fidelity systems have a
bandwidth of 22 khz).
Conversational sound (speech) consists of talk-spurts
followed by silence periods [3], [5], [6]. The ratio is
typically 35:65 respectively, with only one person
speaking at a time.
talk-spurt
350 ms
silence
650 ms
Figure 1: Speech pattern
In digitized voice, the voice signal is sampled at the
Nyquist rate (twice the signal rate) in order to recover the
original signal correctly.
Therefore the voice sampling rate is 2*4 khz = 8 khz or
one sample every 125 ms in telephony. For stereo sound,
the audio frequencies extend up to 22 khz. Hence
sampling frequencies of up to 44 khz are also used in
digital stereo sound. Each sample is coded into a bit-
stream and the number of code-bits varies depending
upon the coding scheme used.
Table 2: Compressed digital audio streams
TRANSMISSION RATES
2400 bps compressed
64 kbps standard PCM
40-16 kbps ADPCM scheme[22]
300-400 kbps stereo
bits/sample
1-2
6-10
5-2
8-10
Although telephone voice is not very bandwidth
intensive, stereo sound requires up to 0.4 Mbps
bandwidth. At lower bit rates, the audio-quality
deteriorates. Determining the quality of digitized audio is
a very subjective phenomenon and studies have indicated
a wide range of acceptable quality. It also depends on the
application. Spoken voice can be compressed
significantly before it becomes incomprehensible. The
study of stereo quality sound is even more subjective.
There are several coding and compression techniques
available for voice [2], [18], [21], [22]. Adaptive
Differential Pulse Code Modulation (ADPCM) and Digital
Speech Interpolation (DSI) are two of the popular
mechanisms for voice compression. DSI is well-suited for
packetized voice transmission as it conserves bandwidth
during silence intervals in a conversation.
4.1: Issues in sound transmissions
In voice transmissions [2],[3],[4],[5],[19], bandwidth is
not an issue in the LAN environment, although it may be
in stereo sound. A critical issue in interactive speech or
music (or voice mail) played back over the network is
latency. Once a person starts speaking, or music starts to
play, intermittent delays during the speech or music
相關(guān)PDF資料 |
PDF描述 |
|---|---|
| CONFIGURE | Configure Low-Cost Buck Controller for Boost Operation (21k) |
| CONFIGURING | Configuring the ?lanSC300 Device＇s Internal CGA Controller for a Specific LCD Panel Application Note? 277KB (PDF) |
| CONTROLINTEGRATEDCIRCUITS | Control Integrated Circuits(22.16 k) |
| COREWARE | CoreWare Merlin2 Fibre Channel core family |
| CORRENT-FS | Smart Networks Alliance Solutions Brief for Corrent Corporation |
相關(guān)代理商/技術(shù)參數(shù) |
參數(shù)描述 |
|---|---|
| CONFIG MODULE 0090 | 制造商:PLX Technology 功能描述:X16 CONFIGURATION MODULE FOR PEX 8648 AND PEX 8632 RDK - Boxed Product (Development Kits) |
| CONFIG MODULE 0091 | 制造商:PLX Technology 功能描述:X8X8 CONFIGURATION MODULE FOR PEX 8648 AND PEX 8632 RDK - Boxed Product (Development Kits) |
| CONFIG MODULE 0092 | 功能描述:CONFIG MODULE 0092 制造商:broadcom limited 系列:- 零件狀態(tài):在售 配件類型:接口板 配套使用產(chǎn)品/相關(guān)產(chǎn)品:- 標(biāo)準(zhǔn)包裝:1 |
| CONFIG MODULE 0093 | 功能描述:CONFIG MODULE 0093 制造商:broadcom limited 系列:- 零件狀態(tài):在售 配件類型:接口板 配套使用產(chǎn)品/相關(guān)產(chǎn)品:- 標(biāo)準(zhǔn)包裝:1 |
| CONFIG MODULE 0094 | 制造商:PLX Technology 功能描述:IP1 CONFIGURATION MODULE FOR PEX 8624, PEX 8616, AND PEX 861 - Boxed Product (Development Kits) |
發(fā)布緊急采購，3分鐘左右您將得到回復(fù)。