ANSI SCTE 24-2 pdf free download
ANSI SCTE 24-2 pdf free download.IPCablecom 1.0 Part 2: Audio Codec Requirements for the Provision of Bi-directional Audio Service over Cable Television Networks Using Cable Modems.
5.2.2.1 Latency Control: Buffering While network jitter and corresponding buffering increase call latency, another source of buffering can be induced by the application as a corrective response to severe packet loss. Although the ultimate solution to additional buffering delay is a pristine network, realistically some packet loss will occur. Accounting for lost packets suggests the need for support concealment or reconstruction of lost data, and in many instances these techniques employ some mechanism of redundant information encoding, temporally shifting and embedding audio frames in the data stream. This not only increases the effective bandwidth requirement, but also creates, in effect, an additional buffer to allow for reassembly, increasing latency. In order to apply certain reconstruction methodologies in an optimal fashion, the application needs accurate data regarding the statistical characteristics of the media stream. Some information is available through real-time control protocol (RTCP) mechanisms, such as a gross measure of packet loss. Additional information, such as burst frequency and predictive time-of- day effects, would improve the potential of the application to make optimal adjustments. Planning for the collection and analysis of this type of network information will allow developers more options in the future, potentially creating applications that will increase network utilization efficiency or quality. 5.2.2.2 Latency Control: Optimal Framing/Packetization As outlined in Section 5.2.1, the loss of audio data frames can have a severe impact on audio quality. The packing of multiple audio frames into a single packet will exacerbate the problem, effectively expanding the loss of one packet into the loss of multiple adjacent audio frames of data. This also increases latency by buffering larger portions of audio samples prior to sending. One way to minimize these effects is to send small packets containing the minimum number of frames.
In the case of on-net and ff-net IP connections, transcoding can be eliminated if all necessary codecs are supported on the client. This is, in fact, impractical but can be optimized statistically if a device supports multiple codecs and can be updated periodically. 5.2.4 Bandwidth Minimization There are two primary mechanisms that client devices may employ to minimize the amount of bandwidth used for their audio/video applications: ●A compressed, low bitrate codec may be applied, thus reducing the bandwidth required. ●A codec may employ some form of variable bitrate transmission. The selection of codecs occurs at the device’ s discretion or via network selection, depending on the protocol employed. Regardless, this takes place after the initial capabilities exchange to determine a compatible codec between endpoints, and assumes that the requested bandwidth is granted by the bandwidth broker element. Variable rate transmission may occur when a codec employs methods resulting in a non- constant bitstream representation of voice data. Voice activity detection (VAD)一silence suppression – is a basic form of variable rate transmission, sending little or no data during speaker silence periods. More advanced variable bitrate encoding (VBR) occurs when a codec dynamically optimizes the compression bitstream.
ANSI SCTE 24-2 pdf download.
