diff --git a/index.bs b/index.bs
index 1e38c5726..4375c7e6b 100644
--- a/index.bs
+++ b/index.bs
@@ -1,12 +1,13 @@
 <pre class=metadata>
-Title: Web Audio API
+Title: Web Audio API 1.1
 Shortname: webaudio
-Level: 1
+Level: 1.1
 Group: audiowg
 Status: ED
 ED: https://webaudio.github.io/web-audio-api/
-TR: https://www.w3.org/TR/webaudio/
+TR: https://www.w3.org/TR/webaudio-11/
 Favicon: favicon.png
+Previous Version: https://www.w3.org/TR/2021/REC-webaudio-20210617/
 Previous Version: https://www.w3.org/TR/2021/CR-webaudio-20210114/
 Previous Version: https://www.w3.org/TR/2020/CR-webaudio-20200611/
 Previous Version: https://www.w3.org/TR/2018/CR-webaudio-20180918/
@@ -97,94 +98,6 @@ window.addEventListener('DOMContentLoaded', () => {
     });
 });
 </script>
-<script>
-// This replaces the contents of "divID" with an unordered list of all the
-// amendment boxes with a prefix of "prefix".  Each item of the list will be of
-// the form "label n", where "label" is the specified label to use via "label".
-// "n" is just a sequential number starting at 1.
-//
-// Example usage where we want a list of all the changes whose id begins with
-// "c2361".
-//
-// (div id="change-list-2361")
-// (/div)
-//
-// And add
-//
-//  ListAmendments("c2361", "Correction", "change-list-2361")
-//
-// to the onload event listener at the bottom of this file.
-//
-// (Replace the "(", ")" with "<", ">", respectively above.  Bikeshed mangles
-// this if we use "<" above.
-
-function ListAmendments(prefix, label, divID) {
-  // Find all the nodes whose id starts with prefix.
-  let nodes = document.querySelectorAll('*[id^="' + prefix + '"]');
-  // Find the div element which will be replaced by the unordered list.
-  let div = document.getElementById(divID);
-
-  // Create the unordered list
-  let text = '<ul>';
-  let index = 1;
-  nodes.forEach(x => {
-    text += '<li><a href="#' + x.id + '">' + label + ' ' + index + '</a></li>';
-    ++index;
-    // Insert buttons for prev and next change
-    InsertButtons(x);
-  });
-  text += '</ul>';
-
-  div.innerHTML = text;
-}
-
-// Search for the class "amendment-buttons", and replace the contents of the div
-// with a set of buttons which link to the previous and next related amendment.
-
-function InsertButtons(node) {
-  let list = node.getElementsByClassName("amendment-buttons");
-
-  // We only add buttons to the first class inside the node.
-  if (list && list.length > 0) {
-    let matches = node.id.match(/([ac]\d+)-(\d+)/);
-    let changeID = matches[1];
-    let changeNum = parseInt(matches[2]);
-    // Create buttons.
-    let text = "";
-    if (changeNum > 1) {
-      // Add "previous" button, only if there is a previous change.
-      text += "<button onclick='location.href=";
-      text += '"#' + changeID + '-' + (changeNum-1) + '"';
-      text += "'>Previous Change</button>";
-    }
-
-    if (document.getElementById(changeID + "-" + (changeNum + 1))) {
-      // Add "next" button only if there is a next change
-      text += "<button onclick='location.href=";
-      text += '"#' + changeID + '-' + (changeNum + 1) + '"';
-      text += "'>Next Change</button>";
-    }
-
-    list[0].innerHTML = text;
-  }
-}
-window.addEventListener('load', (event) => {
-  // Add entries here for each change
-  ListAmendments("c2361", "Proposed Correction", "change-list-2361");
-  ListAmendments("c127", "Proposed Correction", "change-list-127");
-  ListAmendments("c2359", "Proposed Correction", "change-list-2359");
-  ListAmendments("c2373", "Proposed Correction", "change-list-2373");
-  ListAmendments("c2321", "Proposed Correction", "change-list-2321");
-  ListAmendments("c2375", "Proposed Correction", "change-list-2375");
-  ListAmendments("c2475", "Proposed Addition", "change-list-2475");
-  ListAmendments("c2444", "Proposed Addition", "change-list-2444");
-  ListAmendments("c2456", "Proposed Addition", "change-list-2456");
-  ListAmendments("c2400", "Proposed Addition", "change-list-2400");
-  ListAmendments("c2512", "Proposed Addition", "change-list-2512");
-  ListAmendments("c2450", "Proposed Addition", "change-list-2450");
-  ListAmendments("c2567", "Proposed Addition", "change-list-2567");
-});
-</script>
 <style>
 @media (prefers-color-scheme: light) {
 :root {
@@ -796,34 +709,9 @@ media element event task source</a>.
 Additionally, a {{BaseAudioContext}} has several private slots <dfn attribute
 for="BaseAudioContext">[[rendering thread state]]</dfn> and <dfn attribute
 for="BaseAudioContext">[[control thread state]]</dfn> that take values from
-{{AudioContextState}}, and that are both
-<div class="correction proposed" id="c2373-1">
-  <span class="marker">Proposed Correction
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2373">Issue 2373</a>-1.
-  </span>
-  Use new Web IDL buffer primitives
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <del>initialy</del>
-  <ins>initially</ins>
-</div>
-<div class="correction proposed" id="c2450-1">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-1.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <del>
-  set to <code>"suspended"</code>.
-  </del>
-  <ins>
-set to <code>"suspended"</code>, and a private slot <dfn attribute for="BaseAudioContext">
+{{AudioContextState}}, and that are both initially set to <code>"suspended"
+</code>, and a private slot <dfn attribute for="BaseAudioContext">
 [[render quantum size]]</dfn> that is an unsigned integer.
-  </ins>
-</div>
 
 <pre class="idl">
 enum AudioContextState {
@@ -860,15 +748,6 @@ enum AudioContextState {
 </table>
 </div>
 
-<div class="correction proposed" id="c2450-2">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-2.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <ins>
 <pre class="idl">
  enum AudioContextRenderSizeCategory {
     "default",
@@ -904,123 +783,58 @@ enum AudioContextState {
         </tr>
 </table>
 </div>
-</ins>
-</div>
 
-<div class="correction proposed" id="c2450-3">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-3.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <del cite="#c2450-3">
-    <xmp class="idl extract" data-no-idl>
-    callback DecodeErrorCallback = undefined (DOMException error);
-
-    callback DecodeSuccessCallback = undefined (AudioBuffer decodedData);
-
-    [Exposed=Window]
-    interface BaseAudioContext : EventTarget {
-        readonly attribute AudioDestinationNode destination;
-        readonly attribute float sampleRate;
-        readonly attribute double currentTime;
-        readonly attribute AudioListener listener;
-        readonly attribute AudioContextState state;
-        [SameObject, SecureContext]
-        readonly attribute AudioWorklet audioWorklet;
-        attribute EventHandler onstatechange;
-
-        AnalyserNode createAnalyser ();
-        BiquadFilterNode createBiquadFilter ();
-        AudioBuffer createBuffer (unsigned long numberOfChannels,
-                                  unsigned long length,
-                                  float sampleRate);
-        AudioBufferSourceNode createBufferSource ();
-        ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
-        ChannelSplitterNode createChannelSplitter (
-            optional unsigned long numberOfOutputs = 6);
-        ConstantSourceNode createConstantSource ();
-        ConvolverNode createConvolver ();
-        DelayNode createDelay (optional double maxDelayTime = 1.0);
-        DynamicsCompressorNode createDynamicsCompressor ();
-        GainNode createGain ();
-        IIRFilterNode createIIRFilter (sequence<double> feedforward,
-                                       sequence<double> feedback);
-        OscillatorNode createOscillator ();
-        PannerNode createPanner ();
-        PeriodicWave createPeriodicWave (sequence<float> real,
-                                         sequence<float> imag,
-                                         optional PeriodicWaveConstraints constraints = {});
-        ScriptProcessorNode createScriptProcessor(
-            optional unsigned long bufferSize = 0,
-            optional unsigned long numberOfInputChannels = 2,
-            optional unsigned long numberOfOutputChannels = 2);
-        StereoPannerNode createStereoPanner ();
-        WaveShaperNode createWaveShaper ();
-
-        Promise<AudioBuffer> decodeAudioData (
-            ArrayBuffer audioData,
-            optional DecodeSuccessCallback? successCallback,
-            optional DecodeErrorCallback? errorCallback);
-    };
-    </xmp>
-  </del>
-  <ins cite="#c2450-3">
-    <xmp class="idl">
-    callback DecodeErrorCallback = undefined (DOMException error);
-
-    callback DecodeSuccessCallback = undefined (AudioBuffer decodedData);
-
-    [Exposed=Window]
-    interface BaseAudioContext : EventTarget {
-        readonly attribute AudioDestinationNode destination;
-        readonly attribute float sampleRate;
-        readonly attribute double currentTime;
-        readonly attribute AudioListener listener;
-        readonly attribute AudioContextState state;
-        readonly attribute unsigned long renderQuantumSize;
-        [SameObject, SecureContext]
-        readonly attribute AudioWorklet audioWorklet;
-        attribute EventHandler onstatechange;
-
-        AnalyserNode createAnalyser ();
-        BiquadFilterNode createBiquadFilter ();
-        AudioBuffer createBuffer (unsigned long numberOfChannels,
-                                  unsigned long length,
-                                  float sampleRate);
-        AudioBufferSourceNode createBufferSource ();
-        ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
-        ChannelSplitterNode createChannelSplitter (
-            optional unsigned long numberOfOutputs = 6);
-        ConstantSourceNode createConstantSource ();
-        ConvolverNode createConvolver ();
-        DelayNode createDelay (optional double maxDelayTime = 1.0);
-        DynamicsCompressorNode createDynamicsCompressor ();
-        GainNode createGain ();
-        IIRFilterNode createIIRFilter (sequence<double> feedforward,
-                                       sequence<double> feedback);
-        OscillatorNode createOscillator ();
-        PannerNode createPanner ();
-        PeriodicWave createPeriodicWave (sequence<float> real,
-                                         sequence<float> imag,
-                                         optional PeriodicWaveConstraints constraints = {});
-        ScriptProcessorNode createScriptProcessor(
-            optional unsigned long bufferSize = 0,
-            optional unsigned long numberOfInputChannels = 2,
-            optional unsigned long numberOfOutputChannels = 2);
-        StereoPannerNode createStereoPanner ();
-        WaveShaperNode createWaveShaper ();
-
-        Promise<AudioBuffer> decodeAudioData (
-            ArrayBuffer audioData,
-            optional DecodeSuccessCallback? successCallback,
-            optional DecodeErrorCallback? errorCallback);
-    };
-    </xmp>
-  </ins>
-</div>
+<xmp class="idl">
+callback DecodeErrorCallback = undefined (DOMException error);
+
+callback DecodeSuccessCallback = undefined (AudioBuffer decodedData);
+
+[Exposed=Window]
+interface BaseAudioContext : EventTarget {
+    readonly attribute AudioDestinationNode destination;
+    readonly attribute float sampleRate;
+    readonly attribute double currentTime;
+    readonly attribute AudioListener listener;
+    readonly attribute AudioContextState state;
+    readonly attribute unsigned long renderQuantumSize;
+    [SameObject, SecureContext]
+    readonly attribute AudioWorklet audioWorklet;
+    attribute EventHandler onstatechange;
+
+    AnalyserNode createAnalyser ();
+    BiquadFilterNode createBiquadFilter ();
+    AudioBuffer createBuffer (unsigned long numberOfChannels,
+                                unsigned long length,
+                                float sampleRate);
+    AudioBufferSourceNode createBufferSource ();
+    ChannelMergerNode createChannelMerger (optional unsigned long numberOfInputs = 6);
+    ChannelSplitterNode createChannelSplitter (
+        optional unsigned long numberOfOutputs = 6);
+    ConstantSourceNode createConstantSource ();
+    ConvolverNode createConvolver ();
+    DelayNode createDelay (optional double maxDelayTime = 1.0);
+    DynamicsCompressorNode createDynamicsCompressor ();
+    GainNode createGain ();
+    IIRFilterNode createIIRFilter (sequence<double> feedforward,
+                                    sequence<double> feedback);
+    OscillatorNode createOscillator ();
+    PannerNode createPanner ();
+    PeriodicWave createPeriodicWave (sequence<float> real,
+                                        sequence<float> imag,
+                                        optional PeriodicWaveConstraints constraints = {});
+    ScriptProcessorNode createScriptProcessor(
+        optional unsigned long bufferSize = 0,
+        optional unsigned long numberOfInputChannels = 2,
+        optional unsigned long numberOfOutputChannels = 2);
+    StereoPannerNode createStereoPanner ();
+    WaveShaperNode createWaveShaper ();
+
+    Promise<AudioBuffer> decodeAudioData (
+        ArrayBuffer audioData,
+        optional DecodeSuccessCallback? successCallback,
+        optional DecodeErrorCallback? errorCallback);
+};
+</xmp>
 
 <h4 id='BaseAudioContext-attributes'>
 Attributes</h4>
@@ -1112,21 +926,10 @@ Attributes</h4>
         Describes the current state of the {{BaseAudioContext}}.  Getting this
         attribute returns the contents of the {{[[control thread state]]}} slot.
 
-
-<div class="correction proposed" id="c2450-4">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-6.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <ins cite="c2450-4">
     : <dfn>renderQuantumSize</dfn>
     ::
         Getting this attribute returns the value of {{BaseAudioContext/[[render
         quantum size]]}} slot.
-  </ins>
 </dl>
 
 <h4 id="BaseAudioContent-methods">
@@ -1403,21 +1206,14 @@ Methods</h4>
         is via its promise return value, the callback parameters are
         provided for legacy reasons.
 
-        <div class="correction proposed" id="c2321-1">
-            <span class="marker">Candidate Correction Issue 2321:</span>
-            <div class="amendment-buttons">
-                Buttons here
-            </div>
-                Encourage implementation to warn authors in case of a corrupted file. It
-                isn't possible to throw because this would be a breaking change.
+        Encourage implementation to warn authors in case of a corrupted file. It
+        isn't possible to throw because this would be a breaking change.
+        
+        <div class=note>
+            Note: If the compressed audio data byte-stream is corrupted but the
+            decoding can otherwise proceed, implementations are encouraged to warn
+            authors for example via the developer tools.
         </div>
-        <ins cite="#c2321-1">
-            <div class=note>
-                Note: If the compressed audio data byte-stream is corrupted but the
-                decoding can otherwise proceed, implementations are encouraged to warn
-                authors for example via the developer tools.
-            </div>
-        </ins>
 
         <div algorithm="decodeAudioData()">
             <span class="synchronous">When <code>decodeAudioData</code> is
@@ -1428,35 +1224,14 @@ Methods</h4>
 
             2. Let <var>promise</var> be a new Promise.
 
-            3. <div class="correction proposed" id="c2361-1">
-                <span class="marker">Proposed Correction
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-1.
-                </span>
-                Use new Web IDL buffer primitives
-                <div class="amendment-buttons">
-                  Buttons here
-                </div>
-                If <del>
-                the operation <a href="https://tc39.github.io/ecma262/#sec-isdetachedbuffer"><code>IsDetachedBuffer</code></a> (described in [[!ECMASCRIPT]]) on
-                {{BaseAudioContext/decodeAudioData(audioData, successCallback, errorCallback)/audioData!!argument}}
-                is <code>false</code>,</del><ins> {{BaseAudioContext/decodeAudioData(audioData, successCallback, errorCallback)/audioData!!argument}} is [=BufferSource/detached=],</ins> execute the following steps:
-                </div>
+            3. If {{BaseAudioContext/decodeAudioData(audioData, successCallback, errorCallback)/audioData!!argument}} 
+                is [=BufferSource/detached=], execute the following steps:
 
                 1. Append <var>promise</var> to {{BaseAudioContext/[[pending promises]]}}.
 
-                2. <div class="correction proposed" id="c2361-2">
-                    <span class="marker">Proposed Correction
-                    <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-2.
-                    </span>
-                    Use new Web IDL buffer primitives
-                    <div class="amendment-buttons">
-                      Buttons here
-                    </div>
-                    <del><a href="https://tc39.github.io/ecma262/#sec-detacharraybuffer">Detach</a></del><ins>[=ArrayBuffer/Detach=]</ins>
+                2. [=ArrayBuffer/Detach=]
                     the {{BaseAudioContext/decodeAudioData(audioData, successCallback, errorCallback)/audioData!!argument}} {{ArrayBuffer}}.
-                    <del>This operation is described in [[!ECMASCRIPT]].</del>
-                     If this operations throws, jump to the step 3.
-                    </div>
+                    If this operations throws, jump to the step 3. 
 
                 3. Queue a decoding operation to be performed on another thread.
 
@@ -1496,23 +1271,11 @@ Methods</h4>
                 errorCallback)/audioData!!argument}} into [=linear PCM=]. In case of
                 failure, set <var>can decode</var> to <em>false</em>.
 
-                <div class="correction proposed" id="c2375-1">
-                    <span class="marker">Proposed Correction
-                        <a href="https://github.com/WebAudio/web-audio-api/issues/2375">Issue
-                            2375</a>.
-                    </span>
-                    Only decode the first audio track of a multi-track media file.
-                    <div class="amendment-buttons">
-                        Buttons here
-                    </div>
-                    <ins cite=#c2375>
-                        If the media byte-stream contains multiple audio tracks, only decode the
-                        first track to [=linear pcm=].
-
-                        Note: Authors who need more control over the decoding process can use
-                        [[WEBCODECS]].
-                    </ins>
-                </div>
+                If the media byte-stream contains multiple audio tracks, only decode the
+                first track to [=linear pcm=].
+
+                Note: Authors who need more control over the decoding process can use
+                [[WEBCODECS]].
 
             4. If |can decode| is `false`,
                 <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
@@ -1684,12 +1447,6 @@ enum AudioContextLatencyCategory {
 </table>
 </div>
 
-<div class="addition proposed" id="c2400-1">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
 <pre class="idl">
 enum AudioSinkType {
     "none"
@@ -1711,62 +1468,29 @@ enum AudioSinkType {
                     through an audio output device.
     </table>
 </div>
-</ins>
-</div>
 
-<div class="addition proposed" id="c2444-1">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2444">Issue 2444
-        </a> Add AudioRenderCapacity Interface <br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400" id="c2400-2">
-            Issue 2400</a> Access to a different output device <br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2567" id="c2567-1">
-            Issue 2567</a>. Device-related error reporting via AudioContext
-    <div class="amendment-buttons"></div>
-    <del cite="#c2444 #c2400 #c2567">
-        <xmp class="idl extract" data-no-idl>
-        [Exposed=Window]
-        interface AudioContext : BaseAudioContext {
-            constructor (optional AudioContextOptions contextOptions = {});
-            readonly attribute double baseLatency;
-            readonly attribute double outputLatency;
-            AudioTimestamp getOutputTimestamp ();
-            Promise<undefined> resume ();
-            Promise<undefined> suspend ();
-            Promise<undefined> close ();
-            MediaElementAudioSourceNode createMediaElementSource (HTMLMediaElement mediaElement);
-            MediaStreamAudioSourceNode createMediaStreamSource (MediaStream mediaStream);
-            MediaStreamTrackAudioSourceNode createMediaStreamTrackSource (
-                MediaStreamTrack mediaStreamTrack);
-            MediaStreamAudioDestinationNode createMediaStreamDestination ();
-        };
-        </xmp>
-    </del>
-    <ins cite="#c2444 #2400 #2567">
-        <xmp class="idl">
-        [Exposed=Window]
-        interface AudioContext : BaseAudioContext {
-            constructor (optional AudioContextOptions contextOptions = {});
-            readonly attribute double baseLatency;
-            readonly attribute double outputLatency;
-            [SecureContext] readonly attribute (DOMString or AudioSinkInfo) sinkId;
-            [SecureContext] readonly attribute AudioRenderCapacity renderCapacity;
-            attribute EventHandler onsinkchange;
-            attribute EventHandler onerror;
-            AudioTimestamp getOutputTimestamp ();
-            Promise<undefined> resume ();
-            Promise<undefined> suspend ();
-            Promise<undefined> close ();
-            [SecureContext] Promise<undefined> setSinkId ((DOMString or AudioSinkOptions) sinkId);
-            MediaElementAudioSourceNode createMediaElementSource (HTMLMediaElement mediaElement);
-            MediaStreamAudioSourceNode createMediaStreamSource (MediaStream mediaStream);
-            MediaStreamTrackAudioSourceNode createMediaStreamTrackSource (
-                MediaStreamTrack mediaStreamTrack);
-            MediaStreamAudioDestinationNode createMediaStreamDestination ();
-        };
-        </xmp>
-    </ins>
-</div>
+<xmp class="idl">
+[Exposed=Window]
+interface AudioContext : BaseAudioContext {
+    constructor (optional AudioContextOptions contextOptions = {});
+    readonly attribute double baseLatency;
+    readonly attribute double outputLatency;
+    [SecureContext] readonly attribute (DOMString or AudioSinkInfo) sinkId;
+    [SecureContext] readonly attribute AudioRenderCapacity renderCapacity;
+    attribute EventHandler onsinkchange;
+    attribute EventHandler onerror;
+    AudioTimestamp getOutputTimestamp ();
+    Promise<undefined> resume ();
+    Promise<undefined> suspend ();
+    Promise<undefined> close ();
+    [SecureContext] Promise<undefined> setSinkId ((DOMString or AudioSinkOptions) sinkId);
+    MediaElementAudioSourceNode createMediaElementSource (HTMLMediaElement mediaElement);
+    MediaStreamAudioSourceNode createMediaStreamSource (MediaStream mediaStream);
+    MediaStreamTrackAudioSourceNode createMediaStreamTrackSource (
+        MediaStreamTrack mediaStreamTrack);
+    MediaStreamAudioDestinationNode createMediaStreamDestination ();
+};
+</xmp>
 
 An {{AudioContext}} is said to be <dfn>allowed to start</dfn> if the user agent
 allows the context state to transition from "{{AudioContextState/suspended}}" to
@@ -1781,12 +1505,7 @@ and to allow it only when the {{AudioContext}}'s [=relevant global object=] has
     ::
         A boolean flag representing whether the context is suspended by user code.
         The initial value is <code>false</code>.
-<div class="addition proposed" id="c2400-3">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
+
     : <dfn>[[sink ID]]</dfn>
     ::
         A {{DOMString}} or an {{AudioSinkInfo}} representing the identifier
@@ -1798,9 +1517,6 @@ and to allow it only when the {{AudioContext}}'s [=relevant global object=] has
     :: 
         An ordered list to store pending {{Promise}}s created by
         {{AudioContext/resume()}}. It is initially empty.
-
-    </ins>
-</div>
 </dl>
 
 <h4 id="AudioContext-constructors">
@@ -1819,276 +1535,153 @@ Constructors</h4>
             <span class="synchronous">When creating an {{AudioContext}},
             execute these steps:</span>
             
-            <div class="addition proposed" id="c2456-1">
-                <span class="marker">Proposed Addition</span><br>
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2456">
-                    Issue 2456</a> Add a MessagePort to the AudioWorkletGlobalScope <br>
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2400" id="c2400-4">
-                    Issue 2400</a> Access to a different output device <br>
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2450" id="c2450-5">
-                    Issue 2450</a> Allow specifying a different render-quantum-size for an AudioContext or OfflineAudioContext
-                <div class="amendment-buttons"></div>
-                <del cite=#2456>
-                    1. Set a {{[[control thread state]]}} to <code>suspended</code> on the {{AudioContext}}.
-
-                    1. Set a {{[[rendering thread state]]}} to <code>suspended</code> on the {{AudioContext}}.
-
-                    1. Set a private slot called {{[[render quantum size]]}} to
-                        <code>128</code> on the {{AudioContext}}.
+            1. Let |context| be a new {{AudioContext}} object.
 
-                    1. Let <code>[[pending resume promises]]</code> be a
-                        slot on this {{AudioContext}}, that is an initially empty ordered list of
-                        promises.
+            1. Set a {{[[control thread state]]}} to <code>suspended</code> on
+                |context|.
 
-                    1. If <code>contextOptions</code> is given, apply the options:
+            1. Set a {{[[rendering thread state]]}} to <code>suspended</code> on
+                |context|.
 
-                    1. Set the internal latency of this {{AudioContext}}
-                        according to <code>contextOptions.{{AudioContextOptions/latencyHint}}</code>, as described
-                        in {{AudioContextOptions/latencyHint}}.
+            1. Let |messageChannel| be a new {{MessageChannel}}.
 
-                    1. If <code>contextOptions.{{AudioContextOptions/renderSizeHint}}</code>
-                         is specified, execute the following steps:
+            1. Let |controlSidePort| be the value of
+                |messageChannel|'s {{MessageChannel/port1}} attribute.
 
-                         1. If equal to <code>"hardware"</code>, the User-Agent can set
-                                {{[[render quantum size]]}} to a value that's best for the current
-                                setup.
+            1. Let |renderingSidePort| be the value of
+                |messageChannel|'s {{MessageChannel/port2}} attribute.
 
-                         1. If an integer has been passed, the User-Agent can set {{[[render
-                                quantum size]]}} to this integer.
+            1. Let |serializedRenderingSidePort| be the result of
+                [$StructuredSerializeWithTransfer$](|renderingSidePort|,
+                « |renderingSidePort| »).
 
-                    1. If <code>contextOptions.{{AudioContextOptions/sampleRate}}</code> is specified,
-                        set the {{BaseAudioContext/sampleRate}}
-                        of this {{AudioContext}} to this value. Otherwise, use
-                        the sample rate of the default output device. If the
-                        selected sample rate differs from the sample rate of the
-                        output device, this {{AudioContext}} MUST resample the
-                        audio output to match the sample rate of the output device.
+            1. Set this {{BaseAudioContext/audioWorklet}}'s {{AudioWorklet/port}}
+                to |controlSidePort|.
 
-                        Note: If resampling is required, the latency of the
-                        AudioContext may be affected, possibly by a large
-                        amount.
+            1. <a>Queue a control message</a> to <a
+                href="#setting-the-messageport-on-the-scope">set the
+                MessagePort on the AudioContextGlobalScope</a>, with
+                |serializedRenderingSidePort|.
 
-                    1. If the context is <a>allowed to start</a>, send a
-                        <a>control message</a> to start processing.
+            1. If <code>contextOptions</code> is given, perform the following
+                substeps:
 
-                    1. Return this {{AudioContext}} object.
-                </del>
-                <ins cite=#2456>
-                    1. Let |context| be a new {{AudioContext}} object.
+                1. If {{AudioContextOptions/sinkId}} is specified, let |sinkId| be
+                    the value of
+                    <code>contextOptions.{{AudioContextOptions/sinkId}}</code> and
+                    run the following substeps:
 
-                    1. Set a {{[[control thread state]]}} to <code>suspended</code> on
-                        |context|.
+                    1. If both |sinkId| and {{AudioContext/[[sink ID]]}} are a type of
+                        {{DOMString}}, and they are equal to each other, abort these
+                        substeps.
 
-                    1. Set a {{[[rendering thread state]]}} to <code>suspended</code> on
-                        |context|.
-
-                    1. Let |messageChannel| be a new {{MessageChannel}}.
+                    1. If |sinkId| is a type of {{AudioSinkOptions}} and
+                        {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}}, and
+                        {{AudioSinkOptions/type}} in |sinkId| and {{AudioSinkInfo/type}}
+                        in {{AudioContext/[[sink ID]]}} are equal, abort these substeps.
 
-                    1. Let |controlSidePort| be the value of
-                        |messageChannel|'s {{MessageChannel/port1}} attribute.
+                    1. Let |validationResult| be the return value of
+                        <a href="#validating-sink-identifier">sink identifier validation
+                        </a> of |sinkId|.
+    
+                    1. If |validationResult| is a type of {{DOMException}}, throw an
+                        exception with |validationResult| and abort these substeps.
 
-                    1. Let |renderingSidePort| be the value of
-                        |messageChannel|'s {{MessageChannel/port2}} attribute.
+                    1. If |sinkId| is a type of {{DOMString}}, set
+                        {{AudioContext/[[sink ID]]}} to |sinkId| and abort these
+                        substeps.
 
-                    1. Let |serializedRenderingSidePort| be the result of
-                        [$StructuredSerializeWithTransfer$](|renderingSidePort|,
-                        « |renderingSidePort| »).
+                    1. If |sinkId| is a type of {{AudioSinkOptions}}, set
+                        {{AudioContext/[[sink ID]]}} to a new instance of
+                        {{AudioSinkInfo}} created with the value of
+                        {{AudioSinkOptions/type}} of |sinkId|.
 
-                    1. Set this {{BaseAudioContext/audioWorklet}}'s {{AudioWorklet/port}}
-                        to |controlSidePort|.
+                1. Set the internal latency of |context| according to
+                    <code>contextOptions.{{AudioContextOptions/latencyHint}}</code>,
+                    as described in {{AudioContextOptions/latencyHint}}.
 
-                    1. <a>Queue a control message</a> to <a
-                        href="#setting-the-messageport-on-the-scope">set the
-                        MessagePort on the AudioContextGlobalScope</a>, with
-                        |serializedRenderingSidePort|.
+                1. If <code>contextOptions.{{AudioContextOptions/sampleRate}}
+                    </code> is specified, set the {{BaseAudioContext/sampleRate}} of
+                    |context| to this value. Otherwise, follow these substeps:
 
-                    1. If <code>contextOptions</code> is given, perform the following
-                        substeps:
-        
-                        1. If {{AudioContextOptions/sinkId}} is specified, let |sinkId| be
-                            the value of
-                            <code>contextOptions.{{AudioContextOptions/sinkId}}</code> and
-                            run the following substeps:
-
-                            1. If both |sinkId| and {{AudioContext/[[sink ID]]}} are a type of
-                                {{DOMString}}, and they are equal to each other, abort these
-                                substeps.
-
-                            1. If |sinkId| is a type of {{AudioSinkOptions}} and
-                                {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}}, and
-                                {{AudioSinkOptions/type}} in |sinkId| and {{AudioSinkInfo/type}}
-                                in {{AudioContext/[[sink ID]]}} are equal, abort these substeps.
-
-                            1. Let |validationResult| be the return value of
-                                <a href="#validating-sink-identifier">sink identifier validation
-                                </a> of |sinkId|.
-            
-                            1. If |validationResult| is a type of {{DOMException}}, throw an
-                                exception with |validationResult| and abort these substeps.
-        
-                            1. If |sinkId| is a type of {{DOMString}}, set
-                                {{AudioContext/[[sink ID]]}} to |sinkId| and abort these
-                                substeps.
-
-                            1. If |sinkId| is a type of {{AudioSinkOptions}}, set
-                                {{AudioContext/[[sink ID]]}} to a new instance of
-                                {{AudioSinkInfo}} created with the value of
-                                {{AudioSinkOptions/type}} of |sinkId|.
-        
-                        1. Set the internal latency of |context| according to
-                            <code>contextOptions.{{AudioContextOptions/latencyHint}}</code>,
-                            as described in {{AudioContextOptions/latencyHint}}.
+                    1. If |sinkId| is the empty string or a type of
+                        {{AudioSinkOptions}}, use the sample rate of the default output
+                        device. Abort these substeps.
+                    
+                    1. If |sinkId| is a {{DOMString}}, use the sample rate of the
+                        output device identified by |sinkId|. Abort these substeps.
+                    
+                    If <code>contextOptions.{{AudioContextOptions/sampleRate}}</code>
+                    differs from the sample rate of the output device, the user agent
+                    MUST resample the audio output to match the sample rate of the
+                    output device.
 
-                        1. If <code>contextOptions.{{AudioContextOptions/sampleRate}}
-                            </code> is specified, set the {{BaseAudioContext/sampleRate}} of
-                            |context| to this value. Otherwise, follow these substeps:
-        
-                            1. If |sinkId| is the empty string or a type of
-                                {{AudioSinkOptions}}, use the sample rate of the default output
-                                device. Abort these substeps.
-                            
-                            1. If |sinkId| is a {{DOMString}}, use the sample rate of the
-                                output device identified by |sinkId|. Abort these substeps.
-                            
-                            If <code>contextOptions.{{AudioContextOptions/sampleRate}}</code>
-                            differs from the sample rate of the output device, the user agent
-                            MUST resample the audio output to match the sample rate of the
-                            output device.
-        
-                            Note: If resampling is required, the latency of |context| may be
-                            affected, possibly by a large amount.
+                    Note: If resampling is required, the latency of |context| may be
+                    affected, possibly by a large amount.
 
-                    1. If |context| is <a>allowed to start</a>, send a
-                        <a>control message</a> to start processing.
+            1. If |context| is <a>allowed to start</a>, send a
+                <a>control message</a> to start processing.
 
-                    1. Return |context|.
-                </ins>
-            </div>
+            1. Return |context|.
         </div>
 
-
-
     <div algorithm="sending a control message to start processing">
         Sending a <a>control message</a> to start processing means
         executing the following steps:
-    
-        <div class="addition proposed">
-            <div class="addition proposed" id="c2400-5" style="border:0; margin:0; padding:0">
-                <span class="marker">Proposed Addition
-                    <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue 2400</a>.
-                </span>
-                Access to a different output device
-                <div class="amendment-buttons"></div>
-            </div>
-            <div class="addition proposed" id="c2567-2" style="border:0; margin:0; padding:0">
-                <span class="marker">Proposed Addition
-                    <a href="https://github.com/WebAudio/web-audio-api/issues/2567">Issue 2567</a>.
-                </span>
-                Device-related error reporting via AudioContext
-                <div class="amendment-buttons"></div>
-            </div>
-
-            <del cite="#2400 #2567">
-                1. Attempt to <a href="#acquiring">acquire system resources</a>.
-                    In case of failure, abort the following steps.
-
-                1. Set the {{[[rendering thread state]]}} to <code>running</code> on the {{AudioContext}}.
 
-                1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                    queue a media element task</a> to execute the following steps:
+        1. Let |document| be the [=current settings object=]'s [=relevant global object=]'s
+                [=associated Document=].
 
-                    1. Set the {{BaseAudioContext/state}} attribute of the {{AudioContext}} to "{{AudioContextState/running}}".
+        1. Attempt to [=acquire system resources=] to use a following audio output device
+            based on {{AudioContext/[[sink ID]]}} for rendering:
 
-                    1. [=Queue a media element task=] to [=fire an event=] named
-                        {{BaseAudioContext/statechange}} at the {{AudioContext}}.
-
-            </del>
-            <ins cite="#2400 #2567">
-                1. Let |document| be the [=current settings object=]'s [=relevant global object=]'s
-                        [=associated Document=].
-
-                1. Attempt to [=acquire system resources=] to use a following audio output device
-                    based on {{AudioContext/[[sink ID]]}} for rendering:
-
-                    * The default audio output device for the empty string.
-                    * A audio output device identified by {{AudioContext/[[sink ID]]}}.
+            * The default audio output device for the empty string.
+            * A audio output device identified by {{AudioContext/[[sink ID]]}}.
+        
+            1. If resource acquisition fails, execute the following steps:
                 
-                    1. If resource acquisition fails, execute the following steps:
-                        
-                        1. If |document| is not allowed to use the feature identified by
-                            <code>"speaker-selection"</code>, abort these substeps.
-
-                        1. [=Queue a media element task=] to [=fire an event=] named
-                            {{AudioContext/error}} at the {{AudioContext}}, and abort the following
-                            steps.
+                1. If |document| is not allowed to use the feature identified by
+                    <code>"speaker-selection"</code>, abort these substeps.
 
-                1. Set [=this=] {{[[rendering thread state]]}} to <code>running</code> on the
-                    {{AudioContext}}.
+                1. [=Queue a media element task=] to [=fire an event=] named
+                    {{AudioContext/error}} at the {{AudioContext}}, and abort the following
+                    steps.
 
-                1. [=Queue a media element task=] to execute the following steps:
+        1. Set [=this=] {{[[rendering thread state]]}} to <code>running</code> on the
+            {{AudioContext}}.
 
-                    1. Set the {{BaseAudioContext/state}} attribute of the {{AudioContext}}
-                        to "{{AudioContextState/running}}".
+        1. [=Queue a media element task=] to execute the following steps:
 
-                    1. [=fire an event=] named {{BaseAudioContext/statechange}} at the
-                        {{AudioContext}}.
-            </ins>
+            1. Set the {{BaseAudioContext/state}} attribute of the {{AudioContext}}
+                to "{{AudioContextState/running}}".
 
-        </div>
+            1. [=fire an event=] named {{BaseAudioContext/statechange}} at the
+                {{AudioContext}}.
     </div>
 
-        <div class="proposed addition" id="c2567-3">
-            <span class="marker">Proposed Addition
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2567">Issue 2567</a>.
-            </span>
-            Device-related error reporting via AudioContext
-            <div class="amendment-buttons"></div>
-            <del>
-            Note: It is unfortunately not possible to programatically notify authors
-            that the creation of the {{AudioContext}} failed. User-Agents are
-            encouraged to log an informative message if they have access to a logging
-            mechanism, such as a developer tools console.
-            </del>
-            <ins>
-            NOTE: In cases where an {{AudioContext}} is constructed with no arguments and resource
-            acquisition fails, the User-Agent will attempt to silently render the audio graph using
-            a mechanism that emulates an audio output device.
-            </ins>
-        </div>
+    NOTE: In cases where an {{AudioContext}} is constructed with no arguments and resource
+    acquisition fails, the User-Agent will attempt to silently render the audio graph using
+    a mechanism that emulates an audio output device.
 
-        </div>
+    <div algorithm="setting the messageport on the scope" id="setting-the-messageport-on-the-scope">
+        Sending a <a>control message</a> to set the {{MessagePort}} on the
+        {{AudioWorkletGlobalScope}} means executing the following steps, on
+        the <a>rendering thread</a>, with
+        |serializedRenderingSidePort|, that has been transfered to
+        the {{AudioWorkletGlobalScope}}:
 
-        <div class="addition proposed" id="c2456-2">
-            <span class="marker">Proposed Addition
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-                    2456</a>.
-            </span>
-            Add a MessagePort to the AudioWorkletGlobalScope
-            <div class="amendment-buttons">
-            </div>
-            <ins cite=#2456>
-                <div algorithm="setting the messageport on the scope" id="setting-the-messageport-on-the-scope">
-
-                    Sending a <a>control message</a> to set the {{MessagePort}} on the
-                    {{AudioWorkletGlobalScope}} means executing the following steps, on
-                    the <a>rendering thread</a>, with
-                    |serializedRenderingSidePort|, that has been transfered to
-                    the {{AudioWorkletGlobalScope}}:
-
-                    1. Let |deserializedPort| be the result of
-                        [$StructuredDeserialize$](|serializedRenderingSidePort|,
-                        the current Realm).
-
-                    1. Set {{AudioWorkletGlobalScope/port}} to
-                        |deserializedPort|.
+        1. Let |deserializedPort| be the result of
+            [$StructuredDeserialize$](|serializedRenderingSidePort|,
+            the current Realm).
 
-                </div>
-            </ins>
-        </div>
+        1. Set {{AudioWorkletGlobalScope/port}} to
+            |deserializedPort|.
+    </div>
+
+<pre class="argumentdef" for="AudioContext/constructor(contextOptions)">
+contextOptions: User-specified options controlling how the {{AudioContext}} should be constructed.
+</pre>
 
-        <pre class="argumentdef" for="AudioContext/constructor(contextOptions)">
-        contextOptions: User-specified options controlling how the {{AudioContext}} should be constructed.
-        </pre>
 </dl>
 
 <h4 id="AudioContext-attributes">
@@ -2106,29 +1699,11 @@ Attributes</h4>
         specifically does not include any latency incurred the audio
         graph itself.
 
-<div class="correction proposed" id="c2450-6">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-1.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <del>
-    For example, if the audio context is running at 44.1 kHz and
-    the {{AudioDestinationNode}} implements double buffering
-    internally and can process and output audio each <a>render
-    quantum</a>, then the processing latency is \((2\cdot128)/44100
-    = 5.805 \mathrm{ ms}\), approximately.
-  </del>
-  <ins>
-    For example, if the audio context is running at 44.1 kHz with default render
-    quantum size, and the {{AudioDestinationNode}} implements double buffering
-    internally and can process and output audio each <a>render quantum</a>, then
-    the processing latency is \((2\cdot128)/44100 = 5.805 \mathrm{ ms}\),
-    approximately.
-  </ins>
-</div>
+        For example, if the audio context is running at 44.1 kHz with default render
+        quantum size, and the {{AudioDestinationNode}} implements double buffering
+        internally and can process and output audio each <a>render quantum</a>, then
+        the processing latency is \((2\cdot128)/44100 = 5.805 \mathrm{ ms}\),
+        approximately.
 
     : <dfn>outputLatency</dfn>
     ::
@@ -2146,28 +1721,12 @@ Attributes</h4>
         is running or the associated audio output device changes. It is
         useful to query this value frequently when accurate
         synchronization is required.
-<div class="correction proposed" id="c2444-2">
-    <span class="marker">Proposed Addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2444">Issue
-            2444</a>.
-    </span>
-    Add AudioRenderCapacity Interface
-    <div class="amendment-buttons"></div>
-    <ins cite=#c2444>
+
     : <dfn>renderCapacity</dfn>
     ::
         Returns an {{AudioRenderCapacity}} instance associated with
         an {{AudioContext}}.
 
-    </ins>
-</div>
-
-<div class="addition proposed" id="c2400-6">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
     : <dfn>sinkId</dfn>
     ::
         Returns the value of {{AudioContext/[[sink ID]]}} internal slot. This
@@ -2184,16 +1743,6 @@ Attributes</h4>
         construction of {{AudioContext}}. The {{BaseAudioContext/statechange}} event
         is available to check the readiness of the initial output device.
 
-    </ins>
-</div>
-
-<div class="proposed addition" id="c2567-4">
-    <span class="marker">Proposed Addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2567">Issue 2567</a>.
-    </span>
-    Device-related error reporting via AudioContext
-    <div class="amendment-buttons"></div>
-    <ins cite=#2567>
     : <dfn>onerror</dfn>
     ::
         An [=event handler=] for the {{Event}} dispatched from an {{AudioContext}}. The event
@@ -2205,9 +1754,6 @@ Attributes</h4>
                 the context is {{AudioContextState/running}}.
             * When the operating system reports an audio device malfunction.
 
-    </ins>
-</div>
-
 </dl>
 
 <h4 id="AudioContext-methods">
@@ -2572,143 +2118,129 @@ Methods</h4>
             <em>Return type:</em> {{Promise}}&lt;{{undefined}}&gt;
         </div>
 
-<div class="addition proposed" id="c2400-7">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
-        : <dfn>setSinkId((DOMString or AudioSinkOptions) sinkId)</dfn>
-        ::
-            Sets the identifier of an output device. When this method is invoked, the
-            user agent MUST run the following steps:
-
-            <div algorithm="AudioContext.setSinkId()">
-                1. Let |sinkId| be the method's first argument.
+    : <dfn>setSinkId((DOMString or AudioSinkOptions) sinkId)</dfn>
+    ::
+        Sets the identifier of an output device. When this method is invoked, the
+        user agent MUST run the following steps:
 
-                1. If |sinkId| is equal to {{AudioContext/[[sink ID]]}}, return a
-                    promise, resolve it immediately and abort these steps.
+        <div algorithm="AudioContext.setSinkId()">
+            1. Let |sinkId| be the method's first argument.
 
-                1. Let |validationResult| be the return value of
-                    <a href="#validating-sink-identifier">sink identifier validation</a>
-                    of |sinkId|.
-                
-                1. If |validationResult| is not <code>null</code>, return a promise
-                    rejected with |validationResult|. Abort these steps.
+            1. If |sinkId| is equal to {{AudioContext/[[sink ID]]}}, return a
+                promise, resolve it immediately and abort these steps.
 
-                1. Let |p| be a new promise.
+            1. Let |validationResult| be the return value of
+                <a href="#validating-sink-identifier">sink identifier validation</a>
+                of |sinkId|.
+            
+            1. If |validationResult| is not <code>null</code>, return a promise
+                rejected with |validationResult|. Abort these steps.
 
-                1. Send a <a>control message</a> with |p| and |sinkId| to start
-                    processing.
+            1. Let |p| be a new promise.
 
-                1. Return |p|.
-            </div>
+            1. Send a <a>control message</a> with |p| and |sinkId| to start
+                processing.
 
-            <div algorithm="run a control message for setSinkId">
-                Sending a <a>control message</a> to start processing during
-                {{AudioContext/setSinkId()}} means executing the following steps:
+            1. Return |p|.
+        </div>
 
-                1. Let |p| be the promise passed into this algorithm.
+        <div algorithm="run a control message for setSinkId">
+            Sending a <a>control message</a> to start processing during
+            {{AudioContext/setSinkId()}} means executing the following steps:
 
-                1. Let |sinkId| be the sink identifier passed into this algorithm.
+            1. Let |p| be the promise passed into this algorithm.
 
-                1. If both |sinkId| and {{AudioContext/[[sink ID]]}} are a type of
-                    {{DOMString}}, and they are equal to each other,
-                    <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                    queue a media element task</a> to resolve |p| and abort these steps.
+            1. Let |sinkId| be the sink identifier passed into this algorithm.
 
-                1. If |sinkId| is a type of {{AudioSinkOptions}} and
-                    {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}}, and
-                    {{AudioSinkOptions/type}} in |sinkId| and {{AudioSinkInfo/type}} in
-                    {{AudioContext/[[sink ID]]}} are equal,
-                    <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                    queue a media element task</a> to resolve |p| and abort these steps.
+            1. If both |sinkId| and {{AudioContext/[[sink ID]]}} are a type of
+                {{DOMString}}, and they are equal to each other,
+                <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
+                queue a media element task</a> to resolve |p| and abort these steps.
 
-                1. Let |wasRunning| be true.
+            1. If |sinkId| is a type of {{AudioSinkOptions}} and
+                {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}}, and
+                {{AudioSinkOptions/type}} in |sinkId| and {{AudioSinkInfo/type}} in
+                {{AudioContext/[[sink ID]]}} are equal,
+                <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
+                queue a media element task</a> to resolve |p| and abort these steps.
 
-                1. Set |wasRunning| to false if the {{[[rendering thread state]]}} on
-                    the {{AudioContext}} is <code>"suspended"</code>.
+            1. Let |wasRunning| be true.
 
-                1. Pause the renderer after processing the current render quantum.
+            1. Set |wasRunning| to false if the {{[[rendering thread state]]}} on
+                the {{AudioContext}} is <code>"suspended"</code>.
 
-                1. Attempt to <a>release system resources</a>.
+            1. Pause the renderer after processing the current render quantum.
 
-                1. If |wasRunning| is true:
+            1. Attempt to <a>release system resources</a>.
 
-                    1. Set the {{[[rendering thread state]]}} on the {{AudioContext}} to
-                        <code>"suspended"</code>.
-                
-                    1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                        Queue a media element task</a> to execute the following steps:
+            1. If |wasRunning| is true:
 
-                        1. If the {{BaseAudioContext/state}} attribute of the
-                            {{AudioContext}} is not already "{{AudioContextState/suspended}}":
+                1. Set the {{[[rendering thread state]]}} on the {{AudioContext}} to
+                    <code>"suspended"</code>.
+            
+                1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
+                    Queue a media element task</a> to execute the following steps:
 
-                            1. Set the {{BaseAudioContext/state}} attribute of the
-                                {{AudioContext}} to "{{AudioContextState/suspended}}".
+                    1. If the {{BaseAudioContext/state}} attribute of the
+                        {{AudioContext}} is not already "{{AudioContextState/suspended}}":
 
-                            1. [=Fire an event=] named {{BaseAudioContext/statechange}} at the
-                                associated {{AudioContext}}.
+                        1. Set the {{BaseAudioContext/state}} attribute of the
+                            {{AudioContext}} to "{{AudioContextState/suspended}}".
 
-                1. Attempt to <a href="#acquiring">acquire system resources</a> to use
-                    a following audio output device based on {{AudioContext/[[sink ID]]}}
-                    for rendering:
+                        1. [=Fire an event=] named {{BaseAudioContext/statechange}} at the
+                            associated {{AudioContext}}.
 
-                    * The default audio output device for the empty string.
-                    * A audio output device identified by {{AudioContext/[[sink ID]]}}.
-                
-                    In case of failure, reject |p| with "{{InvalidAccessError}}" abort
-                    the following steps.
-                
-                1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                    Queue a media element task</a> to execute the following steps:
+            1. Attempt to <a href="#acquiring">acquire system resources</a> to use
+                a following audio output device based on {{AudioContext/[[sink ID]]}}
+                for rendering:
 
-                    1. If |sinkId| is a type of {{DOMString}}, set
-                        {{AudioContext/[[sink ID]]}} to |sinkId|. Abort these steps.
+                * The default audio output device for the empty string.
+                * A audio output device identified by {{AudioContext/[[sink ID]]}}.
+            
+                In case of failure, reject |p| with "{{InvalidAccessError}}" abort
+                the following steps.
+            
+            1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
+                Queue a media element task</a> to execute the following steps:
 
-                    1. If |sinkId| is a type of {{AudioSinkOptions}} and
-                        {{AudioContext/[[sink ID]]}} is a type of {{DOMString}},
-                        set {{AudioContext/[[sink ID]]}} to a new instance of
-                        {{AudioSinkInfo}} created with the value of
-                        {{AudioSinkOptions/type}} of |sinkId|.
+                1. If |sinkId| is a type of {{DOMString}}, set
+                    {{AudioContext/[[sink ID]]}} to |sinkId|. Abort these steps.
 
-                    1. If |sinkId| is a type of {{AudioSinkOptions}} and
-                        {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}},
-                        set {{AudioSinkInfo/type}} of {{AudioContext/[[sink ID]]}} to
-                        the {{AudioSinkOptions/type}} value of |sinkId|.
+                1. If |sinkId| is a type of {{AudioSinkOptions}} and
+                    {{AudioContext/[[sink ID]]}} is a type of {{DOMString}},
+                    set {{AudioContext/[[sink ID]]}} to a new instance of
+                    {{AudioSinkInfo}} created with the value of
+                    {{AudioSinkOptions/type}} of |sinkId|.
 
-                    1. Resolve |p|.
-                    
-                    1. [=Fire an event=] named {{AudioContext/sinkchange}} at the
-                        associated {{AudioContext}}.
+                1. If |sinkId| is a type of {{AudioSinkOptions}} and
+                    {{AudioContext/[[sink ID]]}} is a type of {{AudioSinkInfo}},
+                    set {{AudioSinkInfo/type}} of {{AudioContext/[[sink ID]]}} to
+                    the {{AudioSinkOptions/type}} value of |sinkId|.
 
-                1. If |wasRunning| is true:
+                1. Resolve |p|.
+                
+                1. [=Fire an event=] named {{AudioContext/sinkchange}} at the
+                    associated {{AudioContext}}.
 
-                    1. Set the {{[[rendering thread state]]}} on the {{AudioContext}} to
-                        <code>"running"</code>.
+            1. If |wasRunning| is true:
 
-                    1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
-                        Queue a media element task</a> to execute the following steps:
+                1. Set the {{[[rendering thread state]]}} on the {{AudioContext}} to
+                    <code>"running"</code>.
 
-                        1. If the {{BaseAudioContext/state}} attribute of the
-                            {{AudioContext}} is not already "{{AudioContextState/running}}":
+                1. <a href="https://html.spec.whatwg.org/multipage/media.html#queue-a-media-element-task">
+                    Queue a media element task</a> to execute the following steps:
 
-                            1. Set the {{BaseAudioContext/state}} attribute of the
-                                {{AudioContext}} to "{{AudioContextState/running}}".
-                            
-                            1. [=Fire an event=] named {{BaseAudioContext/statechange}} at the
-                                associated {{AudioContext}}.
-            </div>
+                    1. If the {{BaseAudioContext/state}} attribute of the
+                        {{AudioContext}} is not already "{{AudioContextState/running}}":
 
-        </ins>
+                        1. Set the {{BaseAudioContext/state}} attribute of the
+                            {{AudioContext}} to "{{AudioContextState/running}}".
+                        
+                        1. [=Fire an event=] named {{BaseAudioContext/statechange}} at the
+                            associated {{AudioContext}}.
+        </div>
 </dl>
 
-<div class="addition proposed" id="c2400-8">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
 <h4 id="validating-sink-identifier">
 Validating {{AudioContext/sinkId}}</h4>
 
@@ -2731,8 +2263,6 @@ This algorithm is used to validate the information provided to modify
 
     1. Return <code>null</code>.
 </div>
-    </ins>
-</div>
 
 <h4 dictionary id="AudioContextOptions">
 {{AudioContextOptions}}</h4>
@@ -2740,22 +2270,6 @@ This algorithm is used to validate the information provided to modify
 The {{AudioContextOptions}} dictionary is used to
 specify user-specified options for an {{AudioContext}}.
 
-<div class="addition proposed" id="c2400-9">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device<br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2450</a> Allow specifying a different render quantum size
-    <div class="amendment-buttons"></div>
-    <del cite="#c2400 #c2450">
-<pre class="idl extract" data-no-idl>
-    dictionary AudioContextOptions {
-        (AudioContextLatencyCategory or double) latencyHint = "interactive";
-        float sampleRate;
-    };
-</pre>
-    </del>
-    <ins cite=#2400>
 <pre class="idl">
     dictionary AudioContextOptions {
         (AudioContextLatencyCategory or double) latencyHint = "interactive";
@@ -2764,8 +2278,6 @@ specify user-specified options for an {{AudioContext}}.
         (AudioContextRenderSizeCategory or unsigned long) renderSizeHint = "default";
     };
 </pre>
-</ins>
-</div>
 
 <h5 id="dictionary-audiocontextoptions-members">
 Dictionary {{AudioContextOptions}} Members</h5>
@@ -2797,30 +2309,11 @@ Dictionary {{AudioContextOptions}} Members</h5>
         specified, the preferred sample rate of the output device for
         this {{AudioContext}} is used.
 
-    <div class="addition proposed" id="c2400-10">
-        <span class="marker">Proposed Addition</span><br>
-            <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-                2400</a> Access to a different output device
-        <div class="amendment-buttons"></div>
-        <ins cite=#2400>
-        : <dfn>sinkId</dfn>
-        ::
-            The identifier or associated information of the audio output device.
-            See {{AudioContext/sinkId}} for more details.
-
-        </ins>
-    </div>
-
+    : <dfn>sinkId</dfn>
+    ::
+        The identifier or associated information of the audio output device.
+        See {{AudioContext/sinkId}} for more details.
 
-<div class="correction proposed" id="c2450-7">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-1.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <ins>
     : <dfn>renderSizeHint</dfn>
     ::
         This allows users to ask for a particular <a>render quantum size</a> when an
@@ -2829,16 +2322,8 @@ Dictionary {{AudioContextOptions}} Members</h5>
         <a>render quantum size</a> if <code>"hardware"</code> is specified.
 
         It is a hint that might not be honored.
-  </ins>
-</div>
 </dl>
 
-<div class="addition proposed" id="c2400-11">
-    <span class="marker">Proposed Addition</span><br>
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue
-            2400</a> Access to a different output device
-    <div class="amendment-buttons"></div>
-    <ins cite=#2400>
 <h4 dictionary id="AudioSinkOptions">
 {{AudioSinkOptions}}</h4>
 
@@ -2882,9 +2367,6 @@ Attributes</h5>
         A value of {{AudioSinkType}} that represents the type of the device.
 </dl>
 
-</ins>
-</div>
-
 <h4 dictionary id="AudioTimestamp">
 {{AudioTimestamp}}</h4>
 
@@ -2911,143 +2393,131 @@ Dictionary {{AudioTimestamp}} Members</h5>
         [[!hr-time-3]]).
 </dl>
 
-<div class="correction proposed" id="c2444-3">
-    <span class="marker">Proposed Addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2444">Issue
-            2444</a>.
-    </span>
-    Add AudioRenderCapacity Interface
-    <div class="amendment-buttons">
-        Buttons here
-    </div>
-    <ins cite=#c2444>
-        <h4 interface lt="audiorenderCapacity" id="AudioRenderCapacity">
-        {{AudioRenderCapacity}}</h4>
-
-        <pre class="idl">
-        [Exposed=Window]
-        interface AudioRenderCapacity : EventTarget {
-            undefined start(optional AudioRenderCapacityOptions options = {});
-                undefined stop();
-                attribute EventHandler onupdate;
-        };
-        </pre>
+<h4 interface lt="audiorenderCapacity" id="AudioRenderCapacity">
+{{AudioRenderCapacity}}</h4>
 
-        This interface provides rendering performance metrics of an
-        {{AudioContext}}. In order to calculate them, the renderer collects a
-        <a>load value</a> per <a>system-level audio callback</a>.
+<pre class="idl">
+[Exposed=Window]
+interface AudioRenderCapacity : EventTarget {
+    undefined start(optional AudioRenderCapacityOptions options = {});
+        undefined stop();
+        attribute EventHandler onupdate;
+};
+</pre>
 
-        <h5 id="AudioRenderCapacity-attributes">
-        Attributes</h5>
+This interface provides rendering performance metrics of an
+{{AudioContext}}. In order to calculate them, the renderer collects a
+<a>load value</a> per <a>system-level audio callback</a>.
 
-        <dl dfn-type=attribute dfn-for="AudioRenderCapacity">
-            : <dfn>onupdate</dfn>
-            ::
-                The event type of this event handler is <dfn event>update</dfn>. Events
-                dispatched to the event handler will use the
-                {{AudioRenderCapacityEvent}} interface.
-        </dl>
+<h5 id="AudioRenderCapacity-attributes">
+Attributes</h5>
 
-        <h5 id="AudioRenderCapacity-methods">
-        Methods</h5>
+<dl dfn-type=attribute dfn-for="AudioRenderCapacity">
+    : <dfn>onupdate</dfn>
+    ::
+        The event type of this event handler is <dfn event>update</dfn>. Events
+        dispatched to the event handler will use the
+        {{AudioRenderCapacityEvent}} interface.
+</dl>
 
-        <dl dfn-type=method dfn-for="AudioRenderCapacity">
-            : <dfn>start(options)</dfn>
-            ::
-                Starts metric collection and analysis. This will repeatedly [=fire an event=] named
-                {{AudioRenderCapacity/update}} at the {{AudioRenderCapacity}}, using
-                {{AudioRenderCapacityEvent}}, with the given update interval in
-                {{AudioRenderCapacityOptions}}.
-            
-            : <dfn>stop()</dfn>
-            ::
-                Stops metric collection and analysis. It also stops dispatching
-                {{AudioRenderCapacity/update}} events.
-        </dl>
+<h5 id="AudioRenderCapacity-methods">
+Methods</h5>
 
-        <h4 dictionary lt="audiorenderCapacityoptions" id="AudioRenderCapacityOptions">
-        {{AudioRenderCapacityOptions}}</h4>
+<dl dfn-type=method dfn-for="AudioRenderCapacity">
+    : <dfn>start(options)</dfn>
+    ::
+        Starts metric collection and analysis. This will repeatedly [=fire an event=] named
+        {{AudioRenderCapacity/update}} at the {{AudioRenderCapacity}}, using
+        {{AudioRenderCapacityEvent}}, with the given update interval in
+        {{AudioRenderCapacityOptions}}.
+    
+    : <dfn>stop()</dfn>
+    ::
+        Stops metric collection and analysis. It also stops dispatching
+        {{AudioRenderCapacity/update}} events.
+</dl>
 
-        The {{AudioRenderCapacityOptions}} dictionary can be used to provide user
-        options for an {{AudioRenderCapacity}}.
+<h4 dictionary lt="audiorenderCapacityoptions" id="AudioRenderCapacityOptions">
+{{AudioRenderCapacityOptions}}</h4>
 
-        <pre class="idl">
-        dictionary AudioRenderCapacityOptions {
-                double updateInterval = 1;
-        };
-        </pre>
+The {{AudioRenderCapacityOptions}} dictionary can be used to provide user
+options for an {{AudioRenderCapacity}}.
 
-        <h5 id="dictionary-AudioRenderCapacityOptions-members">
-        Dictionary {{AudioRenderCapacityOptions}} Members</h5>
+<pre class="idl">
+dictionary AudioRenderCapacityOptions {
+        double updateInterval = 1;
+};
+</pre>
 
-        <dl dfn-type=dict-member dfn-for="AudioRenderCapacityOptions">
-            : <dfn>updateInterval</dfn>
-            ::
-                An update interval (in second) for dispaching
-                {{AudioRenderCapacityEvent}}s. A <a>load value</a> is calculated
-                per <a>system-level audio callback</a>, and multiple load values will
-                be collected over the specified interval period. For example, if
-                the renderer runs at a 48Khz sample rate and the <a>system-level
-                audio callback</a>'s buffer size is 192 frames, 250 load values
-                will be collected over 1 second interval.
-                
-                If the given value is smaller than the duration of
-                the <a>system-level audio callback</a>, {{NotSupportedError}} is
-                thrown.
-        </dl>
+<h5 id="dictionary-AudioRenderCapacityOptions-members">
+Dictionary {{AudioRenderCapacityOptions}} Members</h5>
 
-        <h4 interface lt="audiorenderCapacityevent" id="AudioRenderCapacityEvent">
-        {{AudioRenderCapacityEvent}}</h4>
-
-        <pre class="idl">
-        [Exposed=Window]
-        interface AudioRenderCapacityEvent : Event {
-            constructor (DOMString type, optional AudioRenderCapacityEventInit eventInitDict = {});
-                readonly attribute double timestamp;
-                readonly attribute double averageLoad;
-                readonly attribute double peakLoad;
-                readonly attribute double underrunRatio;
-        };
+<dl dfn-type=dict-member dfn-for="AudioRenderCapacityOptions">
+    : <dfn>updateInterval</dfn>
+    ::
+        An update interval (in second) for dispaching
+        {{AudioRenderCapacityEvent}}s. A <a>load value</a> is calculated
+        per <a>system-level audio callback</a>, and multiple load values will
+        be collected over the specified interval period. For example, if
+        the renderer runs at a 48Khz sample rate and the <a>system-level
+        audio callback</a>'s buffer size is 192 frames, 250 load values
+        will be collected over 1 second interval.
+        
+        If the given value is smaller than the duration of
+        the <a>system-level audio callback</a>, {{NotSupportedError}} is
+        thrown.
+</dl>
 
-        dictionary AudioRenderCapacityEventInit : EventInit {
-            double timestamp = 0;
-            double averageLoad = 0;
-            double peakLoad = 0;
-            double underrunRatio = 0;
-        };
-        </pre>
+<h4 interface lt="audiorenderCapacityevent" id="AudioRenderCapacityEvent">
+{{AudioRenderCapacityEvent}}</h4>
 
-        <h5 id="AudioRenderCapacityEvent-attributes">
-        Attributes</h5>
+<pre class="idl">
+[Exposed=Window]
+interface AudioRenderCapacityEvent : Event {
+    constructor (DOMString type, optional AudioRenderCapacityEventInit eventInitDict = {});
+        readonly attribute double timestamp;
+        readonly attribute double averageLoad;
+        readonly attribute double peakLoad;
+        readonly attribute double underrunRatio;
+};
 
-        <dl dfn-type=attribute dfn-for="AudioRenderCapacityEvent">
-            : <dfn>timestamp</dfn>
-            ::
-                The start time of the data collection period in terms of the
-                associated {{AudioContext}}'s {{BaseAudioContext/currentTime}}.
-            : <dfn>averageLoad</dfn>
-            ::
-                An average of collected load values over the given update
-                interval. The precision is limited to 1/100th.
-            : <dfn>peakLoad</dfn>
-            ::
-                A maximum value from collected load values over the given update
-                interval. The precision is also limited to 1/100th.
-            : <dfn>underrunRatio</dfn>
-            ::
-                A ratio between the number of buffer underruns (when a
-                <a>load value</a> is greater than 1.0) and the total number of
-                <a>system-level audio callback</a>s over the given update interval.
-                
-                Where \(u\) is the number of buffer underruns and \(N\) is the
-                number of <a>system-level audio callback</a>s over the given update
-                interval, the buffer underrun ratio is:
-                - 0.0 if \(u\) = 0.
-                - Otherwise, compute \(u/N\) and take a ceiling value of the
-                    nearest 100th.
-        </dl>
-    </ins>
-</div>
+dictionary AudioRenderCapacityEventInit : EventInit {
+    double timestamp = 0;
+    double averageLoad = 0;
+    double peakLoad = 0;
+    double underrunRatio = 0;
+};
+</pre>
+
+<h5 id="AudioRenderCapacityEvent-attributes">
+Attributes</h5>
+
+<dl dfn-type=attribute dfn-for="AudioRenderCapacityEvent">
+    : <dfn>timestamp</dfn>
+    ::
+        The start time of the data collection period in terms of the
+        associated {{AudioContext}}'s {{BaseAudioContext/currentTime}}.
+    : <dfn>averageLoad</dfn>
+    ::
+        An average of collected load values over the given update
+        interval. The precision is limited to 1/100th.
+    : <dfn>peakLoad</dfn>
+    ::
+        A maximum value from collected load values over the given update
+        interval. The precision is also limited to 1/100th.
+    : <dfn>underrunRatio</dfn>
+    ::
+        A ratio between the number of buffer underruns (when a
+        <a>load value</a> is greater than 1.0) and the total number of
+        <a>system-level audio callback</a>s over the given update interval.
+        
+        Where \(u\) is the number of buffer underruns and \(N\) is the
+        number of <a>system-level audio callback</a>s over the given update
+        interval, the buffer underrun ratio is:
+        - 0.0 if \(u\) = 0.
+        - Otherwise, compute \(u/N\) and take a ceiling value of the
+            nearest 100th.
+</dl>
 
 <!--
  ███████  ████████ ████████ ██       ████ ██    ██ ████████          ███     ██████
@@ -3090,85 +2560,54 @@ Constructors</h4>
     ::
         <div algorithm="OfflineAudioContext.OfflineAudioContext(contextOptions)">
 
-            <div class="addition proposed" id="c2456-3">
-                <span class="marker">Proposed Addition</span>
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-                    2456</a>. Add a MessagePort to the AudioWorkletGlobalScope<br>
-                <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-                    2450</a>. Allow specifying a different render quantum size
-                </span>
-                <div class="amendment-buttons">
-                </div>
-                <del cite="#2456 #2450">
-                    <p>
-                        If the [=current settings object=]'s [=relevant global object=]'s
-                        [=associated Document=] is NOT [=fully active=],
-                        throw an {{InvalidStateError}} and abort these steps.
-                    </p>
-                    Let |c| be a new {{OfflineAudioContext}} object.
-                    Initialize |c| as follows:
-
-                    1. Set the {{[[control thread state]]}} for |c|
-                        to <code>"suspended"</code>.
-
-                    1. Set the {{[[rendering thread state]]}} for
-                        |c| to <code>"suspended"</code>.
-
-                    1. Construct an {{AudioDestinationNode}} with its
-                        {{AudioNode/channelCount}} set to
-                        <code>contextOptions.numberOfChannels</code>.
-                </del>
-                <ins cite="#c2456 c#2450">
-                    <p>
-                        If the [=current settings object=]'s [=relevant global object=]'s
-                        [=associated Document=] is NOT [=fully active=],
-                        throw an {{InvalidStateError}} and abort these steps.
-                    </p>
-                    Let |c| be a new {{OfflineAudioContext}} object.
-                    Initialize |c| as follows:
-
-                    1. Set the {{[[control thread state]]}} for |c|
-                        to <code>"suspended"</code>.
-
-                    1. Set the {{[[rendering thread state]]}} for
-                        |c| to <code>"suspended"</code>.
-
-                    1. Determine the {{[[render quantum size]]}} for this {{OfflineAudioContext}},
-                            based on the value of the {{OfflineAudioContextOptions/renderSizeHint}}:
-
-                        1. If it has the default value of <code>"default"</code> or
-                               <code>"hardware</code>", set the {{[[render quantum size]]}} private
-                               slot to 128.
-
-                        1. Else, if an integer has been passed, the User-Agent can decide to
-                               honour this value by setting it to the {{[[render quantum size]]}}
-                               private slot.
-
-                    1. Construct an {{AudioDestinationNode}} with its
-                        {{AudioNode/channelCount}} set to
-                        <code>contextOptions.numberOfChannels</code>.
-
-                    1. Let |messageChannel| be a new {{MessageChannel}}.
-
-                    1. Let |controlSidePort| be the value of
-                        |messageChannel|'s {{MessageChannel/port1}} attribute.
-
-                    1. Let |renderingSidePort| be the value of
-                        |messageChannel|'s {{MessageChannel/port2}} attribute.
-
-                    1. Let |serializedRenderingSidePort| be the result of
-                        [$StructuredSerializeWithTransfer$](|renderingSidePort|,
-                        « |renderingSidePort| »).
-
-                    1. Set this {{BaseAudioContext/audioWorklet}}'s {{AudioWorklet/port}} to
-                        |controlSidePort|.
-
-                    1. <a>Queue a control message</a> to <a
-                        href="#setting-the-messageport-on-the-scope">set the
-                        MessagePort on the AudioContextGlobalScope</a>, with
-                        |serializedRenderingSidePort|.
-                </ins>
-            </div>
+            <p>
+                If the [=current settings object=]'s [=relevant global object=]'s
+                [=associated Document=] is NOT [=fully active=],
+                throw an {{InvalidStateError}} and abort these steps.
+            </p>
+            Let |c| be a new {{OfflineAudioContext}} object.
+            Initialize |c| as follows:
+
+            1. Set the {{[[control thread state]]}} for |c|
+                to <code>"suspended"</code>.
+
+            1. Set the {{[[rendering thread state]]}} for
+                |c| to <code>"suspended"</code>.
+
+            1. Determine the {{[[render quantum size]]}} for this {{OfflineAudioContext}},
+                    based on the value of the {{OfflineAudioContextOptions/renderSizeHint}}:
+
+                1. If it has the default value of <code>"default"</code> or
+                        <code>"hardware</code>", set the {{[[render quantum size]]}} private
+                        slot to 128.
+
+                1. Else, if an integer has been passed, the User-Agent can decide to
+                        honour this value by setting it to the {{[[render quantum size]]}}
+                        private slot.
+
+            1. Construct an {{AudioDestinationNode}} with its
+                {{AudioNode/channelCount}} set to
+                <code>contextOptions.numberOfChannels</code>.
+
+            1. Let |messageChannel| be a new {{MessageChannel}}.
+
+            1. Let |controlSidePort| be the value of
+                |messageChannel|'s {{MessageChannel/port1}} attribute.
+
+            1. Let |renderingSidePort| be the value of
+                |messageChannel|'s {{MessageChannel/port2}} attribute.
+
+            1. Let |serializedRenderingSidePort| be the result of
+                [$StructuredSerializeWithTransfer$](|renderingSidePort|,
+                « |renderingSidePort| »).
+
+            1. Set this {{BaseAudioContext/audioWorklet}}'s {{AudioWorklet/port}} to
+                |controlSidePort|.
+
+            1. <a>Queue a control message</a> to <a
+                href="#setting-the-messageport-on-the-scope">set the
+                MessagePort on the AudioContextGlobalScope</a>, with
+                |serializedRenderingSidePort|.
         </div>
 
         <pre class=argumentdef for="OfflineAudioContext/constructor(contextOptions)">
@@ -3412,24 +2851,6 @@ Methods</h4>
 This specifies the options to use in constructing an
 {{OfflineAudioContext}}.
 
-<div class="correction proposed" id="c2450-8">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-8.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons">
-    Buttons here
-  </div>
-  <del>
-<pre class="idl extract" data-no-idl>
-dictionary OfflineAudioContextOptions {
-    unsigned long numberOfChannels = 1;
-    required unsigned long length;
-    required float sampleRate;
-};
-</pre>
-  </del>
-  <ins>
 <pre class="idl">
 dictionary OfflineAudioContextOptions {
     unsigned long numberOfChannels = 1;
@@ -3437,9 +2858,6 @@ dictionary OfflineAudioContextOptions {
     required float sampleRate;
     (AudioContextRenderSizeCategory or unsigned long) renderSizeHint = "default";
 };
-</pre>
-  </ins>
-</div>
 </pre>
 
 <h5 id="dictionary-offlineaudiocontextoptions-members">
@@ -3688,20 +3106,11 @@ Methods</h4>
 
     : <dfn>getChannelData(channel)</dfn>
     ::
-        <div class="correction proposed" id="c2361-3">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-3.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
         According to the rules described in <a href="#acquire-the-content">acquire the content</a>
-        either <del>get a reference to</del><ins>allow [=ArrayBufferView/write|writing=] into</ins>
+        either allow [=ArrayBufferView/write|writing=] into
         or [=get a copy of the buffer source|getting a copy of=]
         the bytes stored in {{[[internal data]]}} in a new
         {{Float32Array}}
-        </div>
 
         A {{UnknownError}} may be thrown if the {{[[internal
         data]]}} or the new {{Float32Array}} cannot be
@@ -3736,32 +3145,12 @@ invoker.
     operation occurs on an {{AudioBuffer}}, run the following steps:
 
     
-    1.
-        <div class="correction proposed" id="c2361-4">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-4.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        If <del>the operation <a href="https://tc39.github.io/ecma262/#sec-isdetachedbuffer"><code>IsDetachedBuffer</code></a>
-        on any of the {{AudioBuffer}}'s {{ArrayBuffer}}s</del><ins>any of the {{AudioBuffer}}'s {{ArrayBuffer}}s are
-        [=BufferSource/detached=]</ins>, return `true`, abort these steps, and
+    1. If any of the {{AudioBuffer}}'s {{ArrayBuffer}}s are
+        [=BufferSource/detached=], return `true`, abort these steps, and
         return a zero-length channel data buffer to the invoker.
-        </div>
 
-    2. <div class="correction proposed" id="c2361-5">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-5.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        <del><a href="https://tc39.github.io/ecma262/#sec-detacharraybuffer">Detach</a></del><ins>[=ArrayBuffer/Detach=]</ins> all {{ArrayBuffer}}s for arrays previously returned
+    2. [=ArrayBuffer/Detach=] all {{ArrayBuffer}}s for arrays previously returned
         by {{AudioBuffer/getChannelData()}} on this {{AudioBuffer}}.
-        </div>
 
         Note: Because {{AudioBuffer}} can only be created via
         {{BaseAudioContext/createBuffer()}} or via the {{AudioBuffer}} constructor, this
@@ -4875,17 +4264,8 @@ Methods</h4>
             the original timeline would have had at time \(t_c\).
         </div>
 
-        <div class="correction proposed" id="c127-1">
-            <span class="marker">Candidate Correction Issue 127:</span>
-            <div class="amendment-buttons">
-                Buttons here
-            </div>
-            A {{RangeError}} is only thrown for negative cancelTime values for
-            cancelAndHoldAtTime and cancelScheduledValues.  See <a
-            href="https://github.com/WebAudio/web-audio-api-v2/issues/127">Issue 127</a>
-        </div>
         <pre class=argumentdef for="AudioParam/cancelAndHoldAtTime()">
-            cancelTime: The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the {{AudioContext}}'s {{BaseAudioContext/currentTime}} attribute.  <span class="synchronous">A {{RangeError}} exception MUST be thrown if <code>cancelTime</code> is negative<del cite=#c127-1> or is not a finite number</del>. If {{AudioParam/cancelAndHoldAtTime()/cancelTime}} is less than {{BaseAudioContext/currentTime}}, it is clamped to {{BaseAudioContext/currentTime}}.
+            cancelTime: The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the {{AudioContext}}'s {{BaseAudioContext/currentTime}} attribute.  <span class="synchronous">A {{RangeError}} exception MUST be thrown if <code>cancelTime</code> is negative. If {{AudioParam/cancelAndHoldAtTime()/cancelTime}} is less than {{BaseAudioContext/currentTime}}, it is clamped to {{BaseAudioContext/currentTime}}.
         </pre>
 
         <div>
@@ -4912,7 +4292,7 @@ Methods</h4>
         removed from the timeline.
 
         <pre class=argumentdef for="AudioParam/cancelScheduledValues()">
-            cancelTime: The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the {{AudioContext}}'s {{BaseAudioContext/currentTime}} attribute.  <span class="synchronous">A {{RangeError}} exception MUST be thrown if <code>cancelTime</code> is negative<del cite=#c127-1> or is not a finite number</del>.</span> If <code>cancelTime</code> is less than {{BaseAudioContext/currentTime}}, it is clamped to {{BaseAudioContext/currentTime}}.
+            cancelTime: The time after which any previously scheduled parameter changes will be cancelled. It is a time in the same time coordinate system as the {{AudioContext}}'s {{BaseAudioContext/currentTime}} attribute.  <span class="synchronous">A {{RangeError}} exception MUST be thrown if <code>cancelTime</code> is negative.</span> If <code>cancelTime</code> is less than {{BaseAudioContext/currentTime}}, it is clamped to {{BaseAudioContext/currentTime}}.
         </pre>
 
         <div>
@@ -5571,28 +4951,12 @@ Methods</h4>
 <dl dfn-type=method dfn-for="AnalyserNode">
     : <dfn>getByteFrequencyData(array)</dfn>
     ::
-        <div class="correction proposed" id="c2361-6">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-6.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        <del>Get a
-        <a href="https://heycam.github.io/webidl/#dfn-get-buffer-source-reference">
-        reference to the bytes</a> held by the {{Uint8Array}}
-        passed as an argument.  Copies the <a>current frequency data</a> to those
-        bytes. If the array has fewer elements than the {{frequencyBinCount}}, the
-        excess elements will be dropped. If the array has more elements than the
-        {{frequencyBinCount}}</del><ins>
         [=ArrayBufferView/Write=] the [=current frequency data=] into |array|. If
         |array|'s [=BufferSource/byte length=] is less than {{frequencyBinCount}},
         the excess elements will be dropped. If |array|'s
-        [=BufferSource/byte length=] is greater than the {{frequencyBinCount}}</ins>, the
+        [=BufferSource/byte length=] is greater than the {{frequencyBinCount}}, the
         excess elements will be ignored. The most recent {{AnalyserNode/fftSize}}
         frames are used in computing the frequency data.
-        </div>
 
         If another call to {{AnalyserNode/getByteFrequencyData()}} or
         {{AnalyserNode/getFloatFrequencyData()}} occurs within the same
@@ -5629,27 +4993,12 @@ Methods</h4>
 
     : <dfn>getByteTimeDomainData(array)</dfn>
     ::
-        <div class="correction proposed" id="c2361-7">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-7.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        <del>Get a
-        <a href="https://heycam.github.io/webidl/#dfn-get-buffer-source-reference">
-        reference to the bytes</a> held by the {{Uint8Array}}
-        passed as an argument.  Copies the <a>current time-domain data</a>
-        (waveform data) into those bytes. If the array has fewer elements than the
-        value of {{AnalyserNode/fftSize}}, the excess elements will be dropped. If
-        the array has more elements than {{AnalyserNode/fftSize}},</del><ins>[=ArrayBufferView/Write=] the [=current time-domain data=] (waveform data)
+        [=ArrayBufferView/Write=] the [=current time-domain data=] (waveform data)
         into |array|. If |array|'s [=BufferSource/byte length=] is less than
         {{AnalyserNode/fftSize}}, the excess elements will be dropped. If |array|'s
-        [=BufferSource/byte length=] is greater than the {{AnalyserNode/fftSize}},</ins>
+        [=BufferSource/byte length=] is greater than the {{AnalyserNode/fftSize}},
         the excess elements will be ignored. The most recent
         {{AnalyserNode/fftSize}} frames are used in computing the byte data.
-        </div>
 
         The values stored in the unsigned byte array are computed in
         the following way. Let \(x[k]\) be the time-domain data. Then
@@ -5674,25 +5023,11 @@ Methods</h4>
 
     : <dfn>getFloatFrequencyData(array)</dfn>
     ::
-        <div class="correction proposed" id="c2361-8">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-8.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        <del>Get a
-        <a href="https://heycam.github.io/webidl/#dfn-get-buffer-source-reference">
-        reference to the bytes</a> held by the
-        {{Float32Array}} passed as an argument.  Copies
-        the <a>current frequency data</a> into those bytes.  If the array has
-        fewer elements than the {{frequencyBinCount}},</del><ins>[=ArrayBufferView/Write=] the [=current frequency data=] into |array|. If
+        [=ArrayBufferView/Write=] the [=current frequency data=] into |array|. If
         |array| has fewer elements than the {{frequencyBinCount}}, the excess
         elements will be dropped. If |array| has more elements than the
-        {{frequencyBinCount}},</ins> the excess elements will be ignored. The most recent
+        {{frequencyBinCount}}, the excess elements will be ignored. The most recent
         {{AnalyserNode/fftSize}} frames are used in computing the frequency data.
-        </div>
 
         If another call to {{AnalyserNode/getFloatFrequencyData()}} or
         {{AnalyserNode/getByteFrequencyData()}} occurs within the same
@@ -5712,28 +5047,12 @@ Methods</h4>
 
     : <dfn>getFloatTimeDomainData(array)</dfn>
     ::
-        <div class="correction proposed" id="c2361-9">
-          <span class="marker">Proposed Correction
-          <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>-9.
-          </span>
-          Use new Web IDL buffer primitives
-          <div class="amendment-buttons">
-            Buttons here
-          </div>
-        <del>Get a
-        <a href="https://heycam.github.io/webidl/#dfn-get-buffer-source-reference">
-        reference to the bytes</a> held by the
-        {{Float32Array}} passed as an argument.  Copies the
-        <a>current time-domain data</a> (waveform data) into those bytes. If the
-        array has fewer elements than the value of {{AnalyserNode/fftSize}}, the
-        excess elements will be dropped. If the array has more elements than
-        {{AnalyserNode/fftSize}},</del><ins>[=ArrayBufferView/Write=] the [=current time-domain data=] (waveform data)
+        [=ArrayBufferView/Write=] the [=current time-domain data=] (waveform data)
         into |array|. If |array| has fewer elements than the value of
         {{AnalyserNode/fftSize}}, the excess elements will be dropped. If |array|
-        has more elements than the value of {{AnalyserNode/fftSize}},</ins> the excess
+        has more elements than the value of {{AnalyserNode/fftSize}}, the excess
         elements will be ignored. The most recent {{AnalyserNode/fftSize}} frames
         are written (after downmixing).
-        </div>
 
         <pre class=argumentdef for="AnalyserNode/getFloatTimeDomainData()">
             array: This parameter is where the time-domain sample data will be copied.
@@ -5878,15 +5197,7 @@ In the following, let \(N\) be the value of the
     $$
     </pre>
 
-<div class="addition proposed" id="c2512-1">
-  <span class="marker">Proposed Addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2512">Issue 2512</a>
-  </span>
-  Specify AnalyserNode treatment of non-fininte values.
-  <div class="amendment-buttons"></div>
-  <ins cite=#c2512>
-        * If \(\hat{X}[k]\) is <code>NaN</code>, positive infinity or negative infinity, set \(\hat{X}[k]\) = 0.
-  </ins>
+    * If \(\hat{X}[k]\) is <code>NaN</code>, positive infinity or negative infinity, set \(\hat{X}[k]\) = 0.
 </div>
 
     for \(k = 0, \ldots, N - 1\).
@@ -10985,24 +10296,6 @@ Attributes</h4>
             A value of "{{2x}}" or "{{4x}}" means that the following steps MUST be
             performed:
 
-<div class="correction proposed" id="c2450-9">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-9.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons"></div>
-  <del>
-            1. Up-sample the input samples to 2x or 4x the sample-rate of
-                the {{AudioContext}}. Thus for each <a>render
-                quantum</a>, generate 256 (for 2x) or 512 (for 4x) samples.
-
-            2. Apply the shaping curve.
-
-            3. Down-sample the result back to the sample-rate of the
-                {{AudioContext}}. Thus taking the 256 (or 512)
-                processed samples, generating 128 as the final result.
-  </del>
-  <ins>
             1. Up-sample the input samples to 2x or 4x the sample-rate of
                 the {{AudioContext}}. Thus for each <a>render
                 quantum</a>, generate twice (for 2x) or four times (for 4x) samples.
@@ -11013,8 +10306,6 @@ Attributes</h4>
                 {{AudioContext}}. Thus taking the previously processed samples
                 processed samples, generating a single render quantum worth of
                 samples as the final result.
-  </ins>
-</div>
         </div>
 
         The exact up-sampling and down-sampling filters are not
@@ -11058,46 +10349,28 @@ Dictionary {{WaveShaperOptions}} Members</h5>
 <h3 interface lt="AudioWorklet" id="AudioWorklet">
 The {{AudioWorklet}} Interface</h3>
 
-<div class="addition proposed" id="c2456-4">
-<span class="marker">Proposed Addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-        2456</a>.
-</span>
-Add a MessagePort to the AudioWorkletGlobalScope
-<div class="amendment-buttons">
-</div>
-<del cite=#2456>
-    <pre class="idl extract" data-no-idl>
-[Exposed=Window, SecureContext]
-interface AudioWorklet : Worklet {
-};
-    </pre>
-</del>
-<ins cite=#2456>
-    <pre class="idl">
+<pre class="idl">
 [Exposed=Window, SecureContext]
 interface AudioWorklet : Worklet {
   readonly attribute MessagePort port;
 };
-    </pre>
+</pre>
 
-    <h4 id="AudioWorklet-attributes">Attributes</h4>
+<h4 id="AudioWorklet-attributes">Attributes</h4>
 
-    <dl dfn-type=attribute dfn-for="AudioWorklet">
-        : <dfn>port</dfn>
-        ::
-            A {{MessagePort}} connected to the port on the
-            {{AudioWorkletGlobalScope}}.
-
-            Note: Authors that register an event listener on the
-            <code>"message"</code> event of this {{AudioWorklet/port}} should
-            call {{MessagePort/close}} on either end of the {{MessageChannel}}
-            (either in the {{AudioWorklet}} or the {{AudioWorkletGlobalScope}}
-            side) to allow for resources to be
-            [[html#ports-and-garbage-collection|collected]].
-    </dl>
-</ins>
-</div>
+<dl dfn-type=attribute dfn-for="AudioWorklet">
+    : <dfn>port</dfn>
+    ::
+        A {{MessagePort}} connected to the port on the
+        {{AudioWorkletGlobalScope}}.
+
+        Note: Authors that register an event listener on the
+        <code>"message"</code> event of this {{AudioWorklet/port}} should
+        call {{MessagePort/close}} on either end of the {{MessageChannel}}
+        (either in the {{AudioWorklet}} or the {{AudioWorkletGlobalScope}}
+        side) to allow for resources to be
+        [[html#ports-and-garbage-collection|collected]].
+</dl>
 
 <h4 id="AudioWorklet-concepts">
 Concepts</h4>
@@ -11231,32 +10504,6 @@ Note: An {{AudioWorkletGlobalScope}} is associated with a single
 for that context. This prevents data races from occurring in global
 scope code running within concurrent threads.
 
-<div class="addition proposed" id="c2456-5">
-<span class="marker">Proposed Addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-        2456</a>.
-</span>
-<span class="marker">Proposed addition
-<a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-10.
-</span>
-Add a MessagePort to the AudioWorkletGlobalScope
-<div class="amendment-buttons">
-</div>
-<del cite="#2456 #2450">
-<xmp class="idl extract" data-no-idl>
-callback AudioWorkletProcessorConstructor = AudioWorkletProcessor (object options);
-
-[Global=(Worklet, AudioWorklet), Exposed=AudioWorklet]
-interface AudioWorkletGlobalScope : WorkletGlobalScope {
-    undefined registerProcessor (DOMString name,
-                                 AudioWorkletProcessorConstructor processorCtor);
-    readonly attribute unsigned long long currentFrame;
-    readonly attribute double currentTime;
-    readonly attribute float sampleRate;
-};
-</xmp>
-</del>
-<ins cite="#2456 #2450">
 <pre class="idl">
 callback AudioWorkletProcessorConstructor = AudioWorkletProcessor (object options);
 
@@ -11271,8 +10518,6 @@ interface AudioWorkletGlobalScope : WorkletGlobalScope {
     readonly attribute MessagePort port;
 };
 </pre>
-</ins>
-</div>
 
 <h5 id="AudioWorkletGlobalScope-attributes">
 Attributes</h5>
@@ -11301,27 +10546,16 @@ Attributes</h5>
         The value of the private slot [[render quantum size]] of the associated
         {{BaseAudioContext}}.
 
-    <div class="addition proposed" id="c2456-6">
-    <span class="marker">Proposed Addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue
-            2456</a>.
-    </span>
-    Add a MessagePort to the AudioWorkletGlobalScope
-    <div class="amendment-buttons">
-    </div>
-    <ins cite=#2456>
-        : <dfn>port</dfn>
-        ::
-            A {{MessagePort}} connected to the port on the {{AudioWorklet}}.
-
-            Note: Authors that register an event listener on the <code>"message"</code>
-            event of this {{AudioWorkletGlobalScope/port}} should call
-            {{MessagePort/close}} on either end of the {{MessageChannel}} (either
-            in the {{AudioWorklet}} or the {{AudioWorkletGlobalScope}} side) to
-            allow for resources to be
-            [[html#ports-and-garbage-collection|collected]].
-    </ins>
-    </div>
+    : <dfn>port</dfn>
+    ::
+        A {{MessagePort}} connected to the port on the {{AudioWorklet}}.
+
+        Note: Authors that register an event listener on the <code>"message"</code>
+        event of this {{AudioWorkletGlobalScope/port}} should call
+        {{MessagePort/close}} on either end of the {{MessageChannel}} (either
+        in the {{AudioWorklet}} or the {{AudioWorkletGlobalScope}} side) to
+        allow for resources to be
+        [[html#ports-and-garbage-collection|collected]].
 </dl>
 
 <h5 id="AudioWorkletGlobalScope-methods">
@@ -12266,12 +11500,6 @@ communication (asynchronous) between
 {{AudioWorkletNode}} and
 {{AudioWorkletProcessor}}. This node does not use any output.
 
-<div class="correction" id="c2359">
-  <span class="marker">Candidate Correction
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2359">Issue 2359.</a>
-  </span>
-  Fix typo in code; semi-colon is incorrect.
-</div>
 <pre line-numbers class="example" highlight="js" title="VUMeterNode - Global Scope (vumeternode.js)">
 /* vumeter-node.js: Main global scope */
 
@@ -12282,7 +11510,7 @@ export default class VUMeterNode extends AudioWorkletNode {
             numberOfOutputs: 0,
             channelCount: 1,
             processorOptions: {
-                updateIntervalInMS: updateIntervalInMS || 16.67<del cite=#c2359>;</del>
+                updateIntervalInMS: updateIntervalInMS || 16.67
             }
         });
 
@@ -12525,25 +11753,11 @@ asynchronous sections can't be reordered.
 <h3 id="rendering-loop">
 Rendering an Audio Graph</h3>
 
-<div class="correction proposed" id="c2450-10">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-10.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons"></div>
-  <del>
-Rendering an audio graph is done in blocks of 128 samples-frames. A
-block of 128 samples-frames is called a <a>render quantum</a>, and
-the <a>render quantum size</a> is 128.
-  </del>
-  <ins>
 Audio graph rendering is done in blocks of sample-frames, with the size of each
 block remaining constant for the lifetime of a {{BaseAudioContext}}. The number of
 sample-frames in a block is called <dfn>render quantum size</dfn>, and the block
 itself is called a <dfn>render quantum</dfn>. Its default value is 128, and it can
 be configured by setting {{AudioContextOptions/renderSizeHint}}.
-  </ins>
-</div>
 
 Operations that happen <dfn>atomically</dfn> on a
 given thread can only be executed when no other [=Atomically|atomic=]
@@ -12554,55 +11768,31 @@ The algorithm for rendering a block of audio from a {{BaseAudioContext}}
 <em>Q</em> is comprised of multiple steps and explained in further detail
 in the algorithm of <a href="#rendering-a-graph">rendering a graph</a>.
 
-<div class="correction proposed" id="c2475-1">
-    <span class="marker">Proposed Addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2475">Issue
-            2375</a>.
-    </span>
-    Add definition for various concepts related to system-level audio callbacks
-    <div class="amendment-buttons">
-        Buttons here
-    </div>
-    <ins cite=#c2475>
-        The {{AudioContext}} <a>rendering thread</a> is driven by a
-        <dfn>system-level audio callback</dfn>, that is periodically 
-        at regular intevals. Each call has a <dfn>system-level audio callback buffer
-        size</dfn>, which is a varying number of sample-frames that needs to be
-        computed on time before the next <a>system-level audio callback</a> arrives.
-
-        A <dfn>load value</dfn> is computed for each <a>system-level audio
-        callback</a>, by dividing its execution duration by the <a>system-level
-        audio callback buffer size</a> divided by {{BaseAudioContext/sampleRate}}.
-
-        Ideally the <a>load value</a> is below 1.0, meaning that it took less time
-        to render the audio that it took to play it out.  An <dfn>audio buffer
-        underrun</dfn> happens when this <a>load value</a> is greater than 1.0: the
-        system could not render audio fast enough for real-time.
-
-        <div class=note>
-            The <a>render quantum size</a> for an audio graph is not necessarily a
-            divisor of the <a>system-level audio callback buffer size</a>. This causes
-            increased audio latencies and reduced possible maximum load without
-            <a>audio buffer underrun</a>.
-        </div>
-
-        Note that the concepts of <a>system-level audio callback</a> and <a>load
-        value</a> do not apply to {{OfflineAudioContext}}s.
-    </ins>
-    <del cite=#c2475>
-        <div class="note">
-            In practice, the {{AudioContext}} <a>rendering thread</a> is
-            often running off a system-level audio callback, that executes in
-            an isochronous fashion.
-
-            An {{OfflineAudioContext}} is not required to have a
-            system-level audio callback, but behaves as if it did with the
-            callback happening as soon as the previous callback is
-            finished.
-        </div>
-    </del>
+The {{AudioContext}} <a>rendering thread</a> is driven by a
+<dfn>system-level audio callback</dfn>, that is periodically 
+at regular intevals. Each call has a <dfn>system-level audio callback buffer
+size</dfn>, which is a varying number of sample-frames that needs to be
+computed on time before the next <a>system-level audio callback</a> arrives.
+
+A <dfn>load value</dfn> is computed for each <a>system-level audio
+callback</a>, by dividing its execution duration by the <a>system-level
+audio callback buffer size</a> divided by {{BaseAudioContext/sampleRate}}.
+
+Ideally the <a>load value</a> is below 1.0, meaning that it took less time
+to render the audio that it took to play it out.  An <dfn>audio buffer
+underrun</dfn> happens when this <a>load value</a> is greater than 1.0: the
+system could not render audio fast enough for real-time.
+
+<div class=note>
+    The <a>render quantum size</a> for an audio graph is not necessarily a
+    divisor of the <a>system-level audio callback buffer size</a>. This causes
+    increased audio latencies and reduced possible maximum load without
+    <a>audio buffer underrun</a>.
 </div>
 
+Note that the concepts of <a>system-level audio callback</a> and <a>load
+value</a> do not apply to {{OfflineAudioContext}}s.
+
 The audio callback is also queued as a task in the <a
 href="#control-message-queue">control message queue</a>. The UA MUST perform
 the following algorithms to process render quanta to fulfill such task by
@@ -12891,36 +12081,14 @@ means copying the input data of this {{AudioNode}} for future
 usage.
 
 <dfn>Computing a block of audio</dfn> means
-<div class="addition proposed" id="c2450-11">
-  <span class="marker">Proposed addition
-    <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>-11.
-  </span>
-  Allow specifying a different render quantum size
-  <div class="amendment-buttons"></div>
-  <del>
-running the algorithm for this {{AudioNode}} to produce 128
-sample-frames.
-  </del>
-  <ins>
 running the algorithm for this {{AudioNode}} to produce
 {{BaseAudioContext/[[render quantum size]]}} sample-frames.
-  </ins>
-</div>
 
 <dfn>Processing an input buffer</dfn> means
 running the algorithm for an {{AudioNode}}, using an <a>input
 buffer</a> and the value(s) of the {{AudioParam}}(s) of this
 {{AudioNode}} as the input for this algorithm.
 
-
-<div class="addition proposed" id="c2567-5">
-    <span class="marker">Proposed addition
-        <a href="https://github.com/WebAudio/web-audio-api/issues/2567">Issue 2567</a>.
-    </span>
-    Device-related error reporting via AudioContext
-    <div class="amendment-buttons"></div>
-    <ins>
-
 <h3 id="error-handling-on-a-running-audio-context">
     Handling an error from System Audio Resources on the {{AudioContext}}</h3>
 
@@ -12957,9 +12125,6 @@ The {{AudioContext}} |audioContext| performs the following steps on <a>rendering
 Note: An example of system audio resource errors would be when an external or wireless audio device
     becoming disconnected during the active rendering of the {{AudioContext}}.
 
-    </ins>
-</div>
-
 <h3 id="unloading-a-document">Unloading a document</h3>
     Additional <a href=
     "https://html.spec.whatwg.org/#unloading-document-cleanup-steps">unloading
@@ -14175,274 +13340,6 @@ class Vec3 {
 
 <h2 id="changes">
 Change Log</h2>
-<h3 id="recommendation-changes-1">
-  Changes since Recommendation of 17 Jun 2021
-</h3>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2450">Issue 2450</a>: Allow specifying a different render-quantum-size for an AudioContext or OfflineAudioContext
-    <div id="change-list-2450">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2512">Issue 2512</a>: Specify AnalyserNode treatment of non-fininte values.
-    <div id="change-list-2512">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2456">Issue 2456</a>: Add a MessagePort to the AudioWorkletGlobalScope
-    <div id="change-list-2456">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2444">Issue 2444</a>: Add AudioRenderCapacity interface and related classes
-    <div id="change-list-2444">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2361">Issue 2361</a>: Use new Web IDL buffer primitives
-    <div id="change-list-2361">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api-v2/issue/127">Issue 127</a>: RangeError is thrown only for negative cancelTime
-    <div id="change-list-127">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issue/2359">Issue 2359</a>: Fix a typo in the VUMeterNode example
-    <div id="change-list-2359">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2373">Issue 2373</a>: Fix a typo: initially instead of initialy
-    <div id="change-list-2373">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2321">Issue 2321</a>: Warn about corrupted fils in decodeAudioData.
-    <div id="change-list-2321">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2375">Issue 2375</a>: decodeAudioData only decodes the first track of a multi-track audio file.
-    <div id="change-list-2375">
-    </div>
-  * <a href="https://github.com/WebAudio/web-audio-api/issues/2475">Issue
-        2475</a>: Add definition for various concepts related to system-level audio callbacks
-    <div id="change-list-2475">
-    </div>
-    * <a href="https://github.com/WebAudio/web-audio-api/issues/2400">Issue 2400
-        </a>: Access to a different output device: AudioContext.setSinkId()
-        <div id="change-list-2400"></div>
-
-<h3 id="changes-2021-05-06">
-  Since Proposed Recommendation of 6 May 2021
-</h3>
-  * Styling, status and boilerplate updates for Recommendation
-  * Updated links to RFC2119, High-Resolution Time, and Audio EQ Cookbook
-  * A spelling error was corrected
-
-<h3 id="changes-2021-01-14">
-  Since Candidate Recommendation of 14 January 2021
-</h3>
-<!-- Last updated: 2021-04-30 -->
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2333">PR 2333</a>: Update links to point to W3C versions
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2334">PR 2334</a>: Use bikeshed to link to ErrorEvent
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2331">PR 2331</a>: Add MIMESniff to normative references
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2328">PR 2328</a>: MediaStream must be resampled to match the context sample rate
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2318">PR 2318</a>: Restore empty of pending processor construction data after successful initialization of AudioWorkProcessor#port.
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2317">PR 2317</a>: Standardize h3/h4 interface and dictionary markup
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2312">PR 2312</a>: Rework description of control thread state and rendering thread state
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2311">PR 2311</a>: Adjust the steps to process a context's regular task queue
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2310">PR 2310</a>: OscillatorNode output is mono
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2308">PR 2308</a>: Refine phrasing for "allowed to start"
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2307">PR 2307</a>: Replace "queue a task" with "queue a media element task"
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2306">PR 2306</a>: Move some steps from AudioWorkletProcessor constructor to the instantiation algorithm
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2304">PR 2304</a>: Add required components for ES operations in the rendering loop
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2301">PR 2301</a>: Define when and how regular tasks are processed wrt the processing model
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2286">PR 2286</a>: Clean up ABSN start algorithm
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2277">PR 2277</a>: Fix compression curve diagram
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2273">PR 2273</a>: Clarify units used in threshold & knee value calculations
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2256">PR 2256</a>: ABSN extrapolates the last output
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2250">PR 2250</a>: Use FrozenArray for AudioWorkletProcessor process()
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2298">PR 2298</a>: Bikeshed HTML validation issues
- 
-<h3 id="changes-2020-06-11">
-  Since Candidate Recommendation of 11 June 2020
-</h3>
-<!-- Last updated: 2020-11-13 -->
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2201">PR 2202</a>: Fixed wrong optionality of IIRFilterNode options
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2191">Issue 2191</a>: Restrict sounds beyond normal hearing
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2210">PR 2210</a>: Return rejected promise when the document is not fully active, for operations returning promises
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2191">Issue 2191</a>: Destination of request created by `addModule`
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2213">Issue 2213</a>: The message queue is for message running on the rendering thread.
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2216">Issue 2216</a>: Use inclusive language in the spec
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2219">PR 2219</a>: Update more terminology in images and markdown documents
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2206">Issue 2206</a>: PannerNode.rollOffFactor with "linear" distance model is not clamped to [0, 1] in main browser engines
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2169">Issue 2169</a>: AudioParamDescriptor has member constraints that are redundant
- * <a href="https://github.com/WebAudio/web-audio-api/issues/1457">Issue 1457</a>: [privacy] Exposing data to an origin: fingerprinting
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2061">Issue 2061</a>: Privacy re-review of latest changes
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2225">Issue 2225</a>: Describe "Planar versus interleaved buffers"
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2231">Issue 2231</a>: WaveShaper [[curve set]] not defined
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2240">Issue 2240</a>: Align with Web IDL specification
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2242">Issue 2242</a>: LInk to undefined instead of using `<code>`
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2227">Issue 2227</a>: Clarify buffer.copyToChannel() must be called before source.buffer = buffer else nothing is played
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2253">PR 2253</a>: Fix duplicated IDs for decode callbacks
- * <a href="https://github.com/WebAudio/web-audio-api/issues/2252">Issue 2252</a>: When are promises in "[[pending resume promises]]" resolved?
- * <a href="https://github.com/WebAudio/web-audio-api/pull/2266">PR 2266</a>: Prohibit arbitrary termination of AudioWorkletGlobalScopes
-
-<h3 id="changestart1">
- Since Candidate Recommendation of 18 September 2018
-</h3>
-<!-- Last updated: 2020-04-29 -->
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2193">Issue 2193</a>: Incorrect azimuth comparison in spatialization algorithm
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2197">Issue 2192</a>: Waveshaper curve interpolation algorithm incorrect
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2171">Issue 2171</a>: Allow not having get parameterDescriptors in an AudioWorkletProcessor
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2184">Issue 2184</a>: PannerNode refDistance description unclear
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2165">Issue 2165</a>: AudioScheduledSourceNode start algorithm incomplete
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2155">Issue 2155</a>: Restore changes accidentally reverted in bikeshed conversion
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2154">Issue 2154</a>: Exception for changing channelCountMode on ScriptProcessorNode does not match browsers
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2153">Issue 2153</a>: Exception for changing channelCount on ScriptProcessorNode does not match browsers
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2152">Issue 2152</a>: close() steps don't make sense
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2150">Issue 2150</a>: AudioBufferOptions requires throwing NotFoundError in cases that can't happen
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2149">Issue 2149</a>: MediaStreamAudioSourceNode constructor has weird check for AudioContext
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2148">Issue 2148</a>: IIRFilterOptions description makes impossible demands
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2147">Issue 2147</a>: PeriodicWave constructor examines lengths of things that might not be there
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2113">Issue 2113</a>: BiquadFilter gain lower bound can be lower.
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2096">Issue 2096</a>: Lifetime of pending processor construction data and exceptions in instantiation of AudioWorkletProcessor
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2087">Issue 2087</a>: Minor issues with BiquadFilter AudioParams
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2083">Issue 2083</a>: Missing text in WaveShaperNode?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2082">Issue 2082</a>: WaveShaperNode curve interpolation incomplete
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2074">Issue 2074</a>: Should the AudioWorkletNode constructor invoke the algorithm for initializing an object that inherits from AudioNode?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2073">Issue 2073</a>: Inconsistencies in constructor descriptions and factory method initialization
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2072">Issue 2072</a>: Clarification on `AudioBufferSourceNode` looping, and loop points
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2071">Issue 2071</a>: cancelScheduledValues with setValueCurveAtTime
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2060">Issue 2060</a>: Would it be helpful to restrict use of `AudioWorkletProcessor.port().postMessage()` in order to facilitate garbage collection?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2051">Issue 2051</a>: Update to constructor operations
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2050">Issue 2050</a>: Restore ConvolverNode channel mixing configurability (up to 2 channels)
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2045">Issue 2045</a>: Should the check on `process()` be removed from `AudioWorkletGlobalScope.registerProcessor()`?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2044">Issue 2044</a>: Remove `options` parameter from `AudioWorkletProcessor` constructor WebIDL
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2036">Issue 2036</a>: Remove `options` parameter of `AudioWorkletProcessor` constructor
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2035">Issue 2035</a>: De-duplicate initial value setting on AudioWorkletNode AudioParams
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2027">Issue 2027</a>: Revise "processor construction data" algorithm
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2021">Issue 2021</a>: AudioWorkletProcessor constructor leads to infinite recursion
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2018">Issue 2018</a>: There are still issues with the setup of an AudioWorkletNode's parameters
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2016">Issue 2016</a>: Clarify `parameters` in AudioWorkletProcessor.process()
-* <a href="https://github.com/WebAudio/web-audio-api/issues/2011">Issue 2011</a>: AudioWorkletNodeOptions.processorOptions should not default to null.
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1989">Issue 1989</a>: Please update to Web IDL changes to optional dictionary defaulting
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1984">Issue 1984</a>: Handling of exceptions in audio worklet is not very clear
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1976">Issue 1976</a>: AudioWorkletProcessor's [[node reference]] seems to be write-only
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1972">Issue 1972</a>: parameterDescriptors handling during AudioWorkletNode initialization is probably wrong
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1971">Issue 1971</a>: AudioWorkletNode options serialization is underdefined
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1970">Issue 1970</a>: "active source" flag handling is a weird monkeypatch
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1969">Issue 1969</a>: It would be clearer if the various validation of AudioWorkletNodeOptions were an explicit step or set of steps
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1966">Issue 1966</a>: parameterDescriptors is not looked up by the AudioWorkletProcessor constructor
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1963">Issue 1963</a>: NewTarget check for AudioWorkletProcessor isn't actually possible with a Web IDL constructor
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1947">Issue 1947</a>: Spec is inconsistent about whether parameterDescriptors is an array or an iterable
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1946">Issue 1946</a>: Population of "node name to parameter descriptor map" needs to be defined
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1945">Issue 1945</a>: registerProcessor is doing odd things with threads and JS values
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1943">Issue 1943</a>: Describe how WaveShaperNode shapes the input with the curve
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1935">Issue 1935</a>: length of AudioWorkletProcessor.process() parameter sequences with inactive inputs
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1932">Issue 1932</a>: Make AudioWorkletNode output buffer available for reading
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1925">Issue 1925</a>: front vs forward
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1902">Issue 1902</a>: Mixer Gain Structure section not needed
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1906">Issue 1906</a>: Steps in rendering algorithm
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1905">Issue 1905</a>: Rendering callbacks are observable
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1904">Issue 1904</a>: Strange Note in algorithm for swapping a control message queue
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1903 ">Issue 1903</a>: Funny sentence about priority and latency
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1901">Issue 1901</a>: AudioWorkletNode state property?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1900">Issue 1900</a>: AudioWorkletProcessor NewTarget undefined
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1899">Issue 1899</a>: Missing synchronous markers
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1897">Issue 1897</a>: WaveShaper curve value setter allows multiple sets
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1896">Issue 1896</a>: WaveShaperNode constructor says curve set is initialized to false
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1471">Issue #1471</a>: AudioNode Lifetime section seems to attempt to make garbage collection observable
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1893">Issue #1893</a>: Active processing for Panner/Convolver/ChannelMerger
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1894">Issue #1894</a>: Funny text in PannerNode.orientationX
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1866">Issue #1866</a>: References to garbage collection
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1851">Issue #1851</a>: Parameter values used for BiquadFilterNode::getFrequencyResponse
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1905">Issue #1905</a>: Rendering callbacks are observable
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1879">Issue #1879</a>: ABSN playback algorithm offset
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1882">Issue #1882</a>: Biquad lowpass/highpass Q
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1303">Issue #1303</a>: MediaElementAudioSourceNode information in a funny place
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1896">Issue #1896</a>: WaveShaperNode constructor says curve set is initialized to false
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1897">Issue #1897</a>: WaveShaper curve value setter allows multiple sets.
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1880">Issue #1880</a>: setOrientation description has confusing paragraph
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1855">Issue #1855</a>: createScriptProcessor parameter requirements
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1857">Issue #1857</a>: Fix typos and bad phrasing
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1788">Issue #1788</a>: Unclear what value is returned by AudioParam.value
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1852">Issue #1852</a>: Fix error condition of AudioNode.disconnect(destinationNode, output, input)
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1841">Issue #1841</a>: Recovering from unstable biquad filters?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1777">Issue #1777</a>: Picture of the coordinate system for panner node
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1802">Issue #1802</a>: Clarify interaction between user-invoked suspend and autoplay policy
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1822">Issue #1822</a>: OfflineAudioContext.suspend can suspend before the given time
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1772">Issue #1772</a>: Sorting tracks alphabetically is underspecified
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1797">Issue #1797</a>: Specification is incomplete for AudioNode.connect()
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1805">Issue #1805</a>: Exception ordering on error
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1790">Issue #1790</a>: Automation example chart has an error (reversed function arguments
-* Fix rendering algorithm iteration and cycle breaking
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1719">Issue #1719</a>: channel count changes in filter nodes with tail time
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1563">Issue #1563</a>: Make decodeAudioData more precise
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1481">Issue #1481</a>: Tighten spec on ABSN output channels?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1762">Issue #1762</a>: Setting convolver buffer more than once?
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1758">Issue #1758</a>: Explicitly include time-domain processing code for BiquadFilterNode
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1770">Issue #1770</a>: Link to correct algorithm for StereoPannerNode, mention algorithm is equal-power
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1753">Issue #1753</a>: Have a single `AudioWorkletGlobalScope` per `BaseAudioContext`
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1746">Issue #1746</a>: AnalyserNode: Clarify how much time domain data we're supposed to keep around
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1741">Issue #1741</a>: Sample rate of AudioBuffer
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1745">Issue #1745</a>: Clarify unit of fftSize
-* <a href="https://github.com/WebAudio/web-audio-api/issues/1743">Issue #1743</a>: Missing normative reference to Fetch
-* Use "get a reference to the bytes" algorithm as needed.
-* Specify rules for determining output chanel count.
-* Clarified rendering algorithm for AudioListener.
-    <h3 id="changestart2">
-    Since Working Draft of 19 June 2018
-    </h3>
-* Minor editorial clarifications.
-* Update implementation-report.html.
-* Widen the valid range of detune values so that any value that doesn't cause 2^(d/1200) to overflow is valid.
-* PannerNode constructor throws errors.
-* Rephrase algorithm for setting buffer and curve.
-* Refine startRendering algorithm.
-* Make "queue a task" link to the HTML spec.
-* Specify more precisely, events overlapping with SetValueCurveAtTime.
-* Add implementation report to gh-pages.
-* Honor the given value in `outputChannelCount`.
-* Initialize bufferDuration outside of process() in ABSN algorithm.
-* Rework definition of ABSN output behavior to account for playbackRate’s interaction with the start(…duration) argument.
-* Add mention of video element in ultrasonic attack surface.
-
-<h3 id="changestart3">
-Since Working Draft of 08 December 2015</h3>
-* Add AudioWorklet and related interfaces to support custom nodes. This replaces ScriptProcessorNode, which is now deprecated.
-* Explicitly say what the channel count, mode, and interpretation values are for all source nodes.
-* Specify the behavior of Web Audio when a document is unloaded.
-* Merge the proposed SpatialListener interface into AudioListener.
-* Rework and clean up algorithms for panning and spatialization and define "magic functions".
-* Clarify that AudioBufferSourceNode looping is limited by duration argument to start().
-* Add constructors with options dictionaries for all node types.
-* Clarify parameter automation method behavior and equations. Handle cases where automation methods may interact with each other.
-* Support latency hints and arbitrary sample rates in AudioContext constructor.
-* Clear up ambiguities in definitions of start() and stop() for scheduled sources.
-* Remove automatic dezippering from AudioParam value setters which now equate to setValueAtTime().
-* Specify normative behavior of DynamicsCompressorNode.
-* Specify that AudioParam.value returns the most recent computed value.
-* Permit AudioBufferSourceNode to specify sub-sample start, duration, loopStart and loopEnd. Respecify algorithms to say exactly how looping works in all scenarios, including dynamic and negative playback rates.
-* Harmonized behavior of IIRFilterNode with BiquadFilterNode.
-* Add diagram describing mono-input-to-matrixed-stereo case.
-* Prevent connecting an AudioNode to an AudioParam of a different AudioContext.
-* Added Audioparam cancelAndHoldAtTime
-* Clarify behaviour of AudioParam.cancelScheduledValues().
-* Add playing reference to MediaElementAudioSourceNodes and MediaStreamAudioSourceNodes.
-* Refactor BaseAudioContext interface out of AudioContext, OfflineAudioContext.
-* OfflineAudioContext inherits from BaseAudioContext, not AudioContext.
-* "StereoPanner" replaced with the correct "StereoPannerNode".
-* Support chaining on AudioNode.connect() and AudioParam automation methods.
-* Specify behavior of events following SetTarget events.
-* Reinstate channelCount declaration for AnalyserNode.
-* Specify exponential ramp behavior when previous value is 0.
-* Specify behavior of setValueCurveAtTime parameters.
-* Add spatialListener attribute to AudioContext.
-* Remove section titled "Doppler Shift".
-* Added a list of nodes and reason why they can add latency, in an informative section.
-* Speced nominal ranges, nyquist, and behavior when outside the range.
-* Spec the processing model for the Web Audio API.
-* Merge the SpatialPannerNode into the PannerNode, undeprecating the PannerNode.
-* Merge the SpatialListener into the AudioListener, undeprecating the AudioListener.
-* Added latencyHint(s).
-* Move the constructor from BaseAudioContext to AudioContext where it belongs; BaseAudioContext is not constructible.
-* Specified the Behavior of automations and nominal ranges.
-* The playbackRate is widened to +/- infinity.
-* setValueCurveAtTime is modified so that an implicit call to setValueAtTime is made at the end of the curve duration.
-* Make setting the `value` attribute of an `AudioParam` strictly equivalent of calling setValueAtTime with AudioContext.currentTime.
-* Add new sections for AudioContextOptions and AudioTimestamp.
-* Add constructor for all nodes.
-* Define ConstantSourceNode.
-* Make the WaveShaperNode have a tail time, depending on the oversampling level.
-* Allow collecting MediaStreamAudioSourceNode or MediaElementAudioSourceNode when they won't play ever again.
-* Add a concept of 'allowed to start' and use it when creating an AudioContext and resuming it from resume() (closes #836).
-* Add AudioScheduledSourceNode base class for source nodes.
-* Mark all AudioParams as being k-rate.
 
 <h2 id="acks">
 Acknowledgements</h2>