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Update name, add limits on maximum value and speed. #22

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29 changes: 17 additions & 12 deletions index.html
Original file line number Diff line number Diff line change
Expand Up @@ -131,17 +131,17 @@ <h3>
</p>
<p>
Let <code>RTCRtpReceiver</code> objects have a
<dfn>[[\PlayoutDelayHint]]</dfn> internal slot initially
<dfn>[[\PlayoutDelay]]</dfn> internal slot initially
initialized to <code>null</code>.
</p>
<pre class="idl">partial interface RTCRtpReceiver {
attribute double? playoutDelayHint;
attribute double? playoutDelay;
};</pre>
<section>
<h2>Attributes</h2>
<dl data-link-for="RTCRtpReceiver" data-dfn-for="RTCRtpReceiver"
class="attributes">
<dt><dfn><code>playoutDelayHint</code></dfn> of type <span
<dt><dfn><code>playoutDelay</code></dfn> of type <span
class="idlAttrType">double</span>, nullable</dt>
<dd>
<p>This attribute allows the application to specify a target duration
Expand Down Expand Up @@ -169,10 +169,10 @@ <h2>Attributes</h2>
[[\AssociatedRemoteMediaStreams]]</a> internal slot, then it will
be synchronized with other tracks (for e.g. audio video
synchronization). This means that even if one of the paired tracks
is delayed through [[\PlayoutDelayHint]] then the User Agent
is delayed through [[\PlayoutDelay]] then the User Agent
synchronization mechanism will automatically delay all others
paired tracks. If multiple such paired tracks are delayed through
<a>[[\PlayoutDelayHint]]</a> by different amounts then the largest
<a>[[\PlayoutDelay]]</a> by different amounts then the largest
of those hints will take precedence in synchronization mechanism.
</p>
<p class="note">
Expand All @@ -183,7 +183,7 @@ <h2>Attributes</h2>
jitterBufferEmittedCount</a></code>.
</p>
<p>On getting, this attribute MUST return the value of the
<a>[[\PlayoutDelayHint]]</a> internal slot.</p>
<a>[[\PlayoutDelay]]</a> internal slot.</p>
<p>On setting, the User Agent MUST run the following steps:</p>
<ol>
<li>
Expand All @@ -195,13 +195,12 @@ <h2>Attributes</h2>
<p>Let <var>delay</var> be the argument to the setter.</p>
</li>
<li>
<p>If <var>delay</var> is negative, <a>throw</a> an
<code>TypeError</code> and abort these steps.</p>
<p>If <var>delay</var> is negative or larger than 4 seconds then,
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<a>throw</a> an <code>RangeError</code> and abort these steps.</p>
</li>
<li>
<p>Set the value of <var>receiver</var>'s
<a>[[\PlayoutDelayHint]]</a> internal slot to
<var>delay</var>.</p>
<p>Set the value of <var>receiver</var>'s <a>[[\PlayoutDelay]]</a>
internal slot to <var>delay</var>.</p>
</li>
<li>
<p>In parallel, begin executing the following steps:</p>
Expand All @@ -219,7 +218,7 @@ <h2>Attributes</h2>
If User Agent chooses delay different from requested one,
for e.g. due to network conditions or physical memory
constraints, this is not reflected in the
<a>[[\PlayoutDelayHint]]</a> internal slot.
<a>[[\PlayoutDelay]]</a> internal slot.
</p>
</li>
<li>
Expand All @@ -231,6 +230,12 @@ <h2>Attributes</h2>
<a href="https://tools.ietf.org/html/rfc5888#section-7">
audio/video synchronization</a> or in response to congestion
control.</p>
<p>The acceleration or deceleration rate may vary depending on
network conditions or the type of audio received (e.g. speech
or background noise). It MAY take several seconds to achieve 1
second of buffering but SHOULD not take more than 30 seconds
assuming packets are being received. The speed MAY be
different for audio and video.</p>
<p class="note">
For audio, acceleration and deceleration can be measured
with <code><a href="https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-insertedsamplesfordeceleration">insertedSamplesForDeceleration</a></code>
Expand Down