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Merge pull request #1668 from jan-ivar/nonlive
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RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
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@aboba
aboba committed Nov 30, 2017
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Expand Up @@ -6610,105 +6610,84 @@ <h2>Methods</h2>
</section>
</div>
<p>The <dfn>RTCRtpContributingSource</dfn> and
<dfn>RTCRtpSynchronizationSource</dfn> objects contain information about
a given contributing source (CSRC) or synchronization source (SSRC),
including the most recent time a
<dfn>RTCRtpSynchronizationSource</dfn> dictionaries contain information
about a given contributing source (CSRC) or synchronization source (SSRC)
respectively, including the most recent time a
packet that the source contributed to was played out. The browser MUST
keep information from RTP packets received in the previous 10 seconds.
When the first audio frame contained in an RTP packet is delivered to the
<code><a>RTCRtpReceiver</a></code>'s <code><a>MediaStreamTrack</a></code>
for playout, the user agent MUST queue a task to update the relevant
<code><a>RTCRtpContributingSource</a></code> and
<code><a>RTCRtpSynchronizationSource</a></code> objects based on the
contents of the packet. The
<code><a>RTCRtpSynchronizationSource</a></code> object corresponding to
the SSRC identifier is updated each time, and if the RTP packet contains
CSRC identifiers, then the <code><a>RTCRtpContributingSource</a></code>
objects corresponding to those CSRC identifiers are also updated.</p>
information for the <code><a>RTCRtpContributingSource</a></code> and
<code><a>RTCRtpSynchronizationSource</a></code> dictionaries based on the
contents of the packet. The information relevant to the
<code><a>RTCRtpSynchronizationSource</a></code> dictionary corresponding
to the SSRC identifier, is updated each time, and if the RTP packet
contains CSRC identifiers, then the information relevant to the
<code><a>RTCRtpContributingSource</a></code> dictionaries corresponding to
those CSRC identifiers is also updated.</p>
<div class="note">As stated in the <a href="#conformance">conformance
section</a>, requirements phrased as algorithms may be implemented in
any manner so long as the end result is equivalent. So, an
implementaion does not need to literally queue a task for every
packet, as long as the end result is that within a single event loop task
execution, all <code><a>RTCRtpSynchronizationSource</a></code> and
<code><a>RTCRtpContributingSource</a></code> objects for a particular
<code><a>RTCRtpReceiver</a></code> return information from a single point
in the RTP stream.</div>
execution, all returned <code><a>RTCRtpSynchronizationSource</a></code>
and <code><a>RTCRtpContributingSource</a></code> dictionaries for a
particular <code><a>RTCRtpReceiver</a></code> contain information from a
single point in the RTP stream.</div>
<div>
<pre class="idl">[Exposed=Window] interface RTCRtpContributingSource {
readonly attribute DOMHighResTimeStamp timestamp;
readonly attribute unsigned long source;
readonly attribute byte? audioLevel;
<pre class="idl">dictionary RTCRtpContributingSource {
required DOMHighResTimeStamp timestamp;
required unsigned long source;
required byte? audioLevel;
};</pre>
<section>
<h2>Attributes</h2>
<h2>Dictionary RTCRtpContributingSource Members</h2>
<dl data-link-for="RTCRtpContributingSource" data-dfn-for=
"RTCRtpContributingSource" class="attributes">
"RTCRtpContributingSource" class="dictionary-members">
<dt><dfn><code>timestamp</code></dfn> of type <span class=
"idlAttrType"><a>DOMHighResTimeStamp</a></span>, readonly</dt>
"idlMemberType"><a>DOMHighResTimeStamp</a></span>, required</dt>
<dd>
<p>The timestamp of type DOMHighResTimeStamp [[!HIGHRES-TIME]],
indicating the most recent time of playout of an RTP packet
containing the source. The timestamp is defined in
[[!HIGHRES-TIME]] and corresponds to a local clock.</p>
</dd>
<dt><dfn><code>source</code></dfn> of type <span class=
"idlAttrType"><a>unsigned long</a></span>, readonly</dt>
"idlMemberType"><a>unsigned long</a></span>, required</dt>
<dd>
<p>The CSRC identifier of the contributing source.</p>
<p>The CSRC or SSRC identifier of the contributing or
synchronization source.</p>
</dd>
<dt><dfn><code>audioLevel</code></dfn> of type <span class=
"idlAttrType"><a>byte</a></span>, readonly , nullable</dt>
"idlMemberType"><a>byte</a></span>, required, nullable</dt>
<dd>
<p>The audio level contained in the last RTP packet played from
this source. <code>audioLevel</code> will be the level value
defined in [[!RFC6465]] if the RFC 6465 header extension is
present, and otherwise <code>null</code>. RFC 6465 defines the
level as a integral value from 0 to 127 representing the audio
level in negative decibels relative to the loudest signal that
the system could possibly encode. Thus, 0 represents the loudest
signal the system could possibly encode, and 127 represents
silence.</p>
this source. For CSRCs, <code>audioLevel</code> must be the level
value defined in [[!RFC6465]] if the RFC 6465 header extension is
present, otherwise <code>null</code>. For SSRCs, it must be the
level value defined in [[!RFC6464]] if the RFC 6464 header
extension is present, otherwise a browser computed value as if it
had come from RFC 6464 (never <code>null</code>).</p>
<p>Both RFCs define the level as an integral value from 0 to 127
representing the audio level in negative decibels relative to the
loudest signal that the system could possibly encode. Thus,
0 represents the loudest signal the system could possibly encode,
and 127 represents silence.</p>
</dd>
</dl>
</section>
</div>
<div>
<pre class="idl">[Exposed=Window] interface RTCRtpSynchronizationSource {
readonly attribute DOMHighResTimeStamp timestamp;
readonly attribute unsigned long source;
readonly attribute byte audioLevel;
readonly attribute boolean? voiceActivityFlag;
<pre class="idl">dictionary RTCRtpSynchronizationSource : RTCRtpContributingSource {
required boolean? voiceActivityFlag;
};</pre>
<section>
<h2>Attributes</h2>
<h2>Dictionary RTCRtpSynchronizationSource Members</h2>
<dl data-link-for="RTCRtpSynchronizationSource" data-dfn-for=
"RTCRtpSynchronizationSource" class="attributes">
<dt><dfn><code>timestamp</code></dfn> of type <span class=
"idlAttrType"><a>DOMHighResTimeStamp</a></span>, readonly</dt>
<dd>
<p>The timestamp of type DOMHighResTimeStamp [[!HIGHRES-TIME]],
indicating the most recent time of playout of an RTP packet
from the source. The timestamp is defined in
[[!HIGHRES-TIME]] and corresponds to a local clock.</p>
</dd>
<dt><dfn><code>source</code></dfn> of type <span class=
"idlAttrType"><a>unsigned long</a></span>, readonly</dt>
<dd>
<p>The SSRC identifier of the synchronization source.</p>
</dd>
<dt><dfn><code>audioLevel</code></dfn> of type <span class=
"idlAttrType"><a>byte</a></span>, readonly, nullable</dt>
<dd>
<p>The audio level contained in the last RTP packet played from
this source. <code>audioLevel</code> will be the level value
defined in [[!RFC6464]], if the RFC 6464 header extension is
present. If the RFC 6464 extension header is not present, the
browser will compute a value for <code>audioLevel</code> as if it
had come from RFC 6464.
</dd>
"RTCRtpSynchronizationSource" class="dictionary-members">
<dt><dfn><code>voiceActivityFlag</code></dfn> of type <span class=
"idlAttrType"><a>boolean</a></span>, readonly, nullable</dt>
"idlMemberType"><a>boolean</a></span>, required, nullable</dt>
<dd>
<p>Whether the last RTP packet played from this source contains
voice activity (true) or not (false). If the RFC 6464 extension
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