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Understanding the Receive Side

Figure 7-10 shows the receiver-side processing. The audio path consists of the jitter buffer, followed by the audio decoder, followed by the digital-to-analog (D/A) converter. The video path consists of a video decoder, a video buffer, and a video playout device.

Figure 7-10

Figure 7-10 Receiver-Side Processing

Audio Receiver Path

The receiver requires the jitter buffer in the audio path because packets arriving at the receiver do not have uniform arrival times. The sending endpoint typically sends fixed-sized RTP packets onto the network at uniform intervals, generating a stream with a constant audio bit rate. However, jitter in the network due to transient delays causes nonuniform spacing between packet arrival times at the receiver. If the network imposes a temporary delay on a sequence of several packets, those packets arrive late, causing the jitter buffer on the receive side to decrease. The jitter buffer must be large enough to prevent the buffer from dropping to the point where it underflows. If the jitter buffer underflows, the audio device has no data to play out the audio speakers, and the user hears a glitch.

This scenario, in which the jitter buffer runs out of data for the audio playout device, is called audio starvation. Conversely, if the network then transfers these delayed packets in quick succession, the burst of packets causes the jitter buffer to rise back to its normal level quickly.

The jitter buffer absorbs these arrival-time variations; however, the jitter buffer imposes an additional delay in the end-to-end audio pipeline. This delay is equal to the average level of the jitter buffer, measured in milliseconds. Therefore, the goal of the receive endpoint is to establish a jitter buffer with the smallest average latency, which can minimize the probability of an audio packet dropout. The endpoint typically adapts the level of the jitter buffer over time by observing the recent history of jitter and increasing the average buffer level if necessary. In fact, if the jitter buffer underflows and results in a dropped packet, the receiver immediately reestablishes a new jitter buffer with a higher average level to accommodate greater variance.

When the jitter buffer underflows, the audio decoder must come to the rescue and supply a replacement for the missing audio packet. This replacement packet may contain audio silence, or it may contain audio that attempts to conceal the lost packet. Packet loss concealment (PLC) is the process of mitigating the loss of quality resulting from a lost packet. One common form of PLC is to just replay the most recent packet received from the network.

The series of audio processing units—including the input buffer, decoder, and playout device—can be considered a data pipeline, each with its own delay. To establish the initial jitter buffer level, the receiver must "fill the pipe" by filling the entire pipeline on the receive side until the audio "backs up" the pipeline to the input buffers and achieves the desired input buffer level.

In addition, the jitter buffer can provide the delay necessary to re-sort out-of-order packets.

The audio decode delay is analogous to the corresponding audio encoding delay on the sending side. The audio hardware playout delay on the receiver is analogous to the audio hardware capture delay on the sender.

Figure 7-11 shows a graphical depiction of the delays on the receive side. When the receiver depacketizes a large packet into smaller packets, no delay results. The reason is because the receiver does not need to wait for successive packets of data to arrive, because depacketization does not perform an aggregation process. Such is the typical case when the receiver depacketizes the RTP packet into audio frames, and again when the decoded audio goes through the depacketization process to be sliced into yet smaller audio device packets for the audio hardware device.

Figure 7-11

Figure 7-11 Receive-Side Audio Processing Delays

Receiver Video Path

The receiver has several delays in the video path:

  • The packetization delay—This latency might be required if the video decoder needs access to more than one slice (or group of blocks) to start the decoding process. However, video conferencing endpoints typically use a low-latency bitstream that allows endpoints to decode a slice without needing to use information from other slices. In this case, the input video packetization process simply reformats the video packet and does not perform any type of packet aggregation, and therefore, this packetization process imposes no delay on the video path.
  • The decode delay—Analogous to the audio decode delay, it reconstructs slices of the video frame.
  • The synchronization delay—If necessary, the receiver may impose a delay on the video frames to achieve synchronization.
  • The playout delay—After the endpoint writes a new decoded video frame into memory, the playout delay is the time until that frame displays on the screen.

Types of Playout Devices

Playout devices come in two types: malleable and nonmalleable. Malleable playout devices can play a media sample on command, at any time. An example of a malleable playout device is a video display monitor. Typically, malleable devices do not request data; for instance, a receiver can send a video frame directly to the display device, and the device immediately writes the frame into video memory. The video frame appears on the screen the next time the TV raster scans the screen.

In contrast, nonmalleable devices always consume data at a constant rate. The audio playout device is an example: The receiver must move data to the audio device at exactly the real-time rate. Nonmalleable devices typically issue interrupt requests each time they must receive new data, and the receiver must service the interrupt request quickly to maintain a constant data rate to the device. After the receiver sends the first packet of audio to the audio device, the audio device typically proceeds to generate interrupt requests on a regular basis to acquire a constant stream of audio data. Audio devices generally receive fixed-size audio device packets of data at each interrupt, and professional audio interfaces can support buffer sizes as low as 64 samples. At 44.1 kHz and a 64-sample buffer size, packets will be 1.5 ms, and the audio device will generate about 689 interrupt requests per second.

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