<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.1//EN"
"http://www.w3.org/TR/xhtml11/DTD/xhtml11.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head><meta http-equiv="content-type" content="text/html; charset=utf-8" />
<title>[213983] trunk</title>
</head>
<body>
<style type="text/css"><!--
#msg dl.meta { border: 1px #006 solid; background: #369; padding: 6px; color: #fff; }
#msg dl.meta dt { float: left; width: 6em; font-weight: bold; }
#msg dt:after { content:':';}
#msg dl, #msg dt, #msg ul, #msg li, #header, #footer, #logmsg { font-family: verdana,arial,helvetica,sans-serif; font-size: 10pt; }
#msg dl a { font-weight: bold}
#msg dl a:link { color:#fc3; }
#msg dl a:active { color:#ff0; }
#msg dl a:visited { color:#cc6; }
h3 { font-family: verdana,arial,helvetica,sans-serif; font-size: 10pt; font-weight: bold; }
#msg pre { overflow: auto; background: #ffc; border: 1px #fa0 solid; padding: 6px; }
#logmsg { background: #ffc; border: 1px #fa0 solid; padding: 1em 1em 0 1em; }
#logmsg p, #logmsg pre, #logmsg blockquote { margin: 0 0 1em 0; }
#logmsg p, #logmsg li, #logmsg dt, #logmsg dd { line-height: 14pt; }
#logmsg h1, #logmsg h2, #logmsg h3, #logmsg h4, #logmsg h5, #logmsg h6 { margin: .5em 0; }
#logmsg h1:first-child, #logmsg h2:first-child, #logmsg h3:first-child, #logmsg h4:first-child, #logmsg h5:first-child, #logmsg h6:first-child { margin-top: 0; }
#logmsg ul, #logmsg ol { padding: 0; list-style-position: inside; margin: 0 0 0 1em; }
#logmsg ul { text-indent: -1em; padding-left: 1em; }#logmsg ol { text-indent: -1.5em; padding-left: 1.5em; }
#logmsg > ul, #logmsg > ol { margin: 0 0 1em 0; }
#logmsg pre { background: #eee; padding: 1em; }
#logmsg blockquote { border: 1px solid #fa0; border-left-width: 10px; padding: 1em 1em 0 1em; background: white;}
#logmsg dl { margin: 0; }
#logmsg dt { font-weight: bold; }
#logmsg dd { margin: 0; padding: 0 0 0.5em 0; }
#logmsg dd:before { content:'\00bb';}
#logmsg table { border-spacing: 0px; border-collapse: collapse; border-top: 4px solid #fa0; border-bottom: 1px solid #fa0; background: #fff; }
#logmsg table th { text-align: left; font-weight: normal; padding: 0.2em 0.5em; border-top: 1px dotted #fa0; }
#logmsg table td { text-align: right; border-top: 1px dotted #fa0; padding: 0.2em 0.5em; }
#logmsg table thead th { text-align: center; border-bottom: 1px solid #fa0; }
#logmsg table th.Corner { text-align: left; }
#logmsg hr { border: none 0; border-top: 2px dashed #fa0; height: 1px; }
#header, #footer { color: #fff; background: #636; border: 1px #300 solid; padding: 6px; }
#patch { width: 100%; }
#patch h4 {font-family: verdana,arial,helvetica,sans-serif;font-size:10pt;padding:8px;background:#369;color:#fff;margin:0;}
#patch .propset h4, #patch .binary h4 {margin:0;}
#patch pre {padding:0;line-height:1.2em;margin:0;}
#patch .diff {width:100%;background:#eee;padding: 0 0 10px 0;overflow:auto;}
#patch .propset .diff, #patch .binary .diff {padding:10px 0;}
#patch span {display:block;padding:0 10px;}
#patch .modfile, #patch .addfile, #patch .delfile, #patch .propset, #patch .binary, #patch .copfile {border:1px solid #ccc;margin:10px 0;}
#patch ins {background:#dfd;text-decoration:none;display:block;padding:0 10px;}
#patch del {background:#fdd;text-decoration:none;display:block;padding:0 10px;}
#patch .lines, .info {color:#888;background:#fff;}
--></style>
<div id="msg">
<dl class="meta">
<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/213983">213983</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2017-03-15 09:38:10 -0700 (Wed, 15 Mar 2017)</dd>
</dl>
<h3>Log Message</h3>
<pre>run-webkit-tests is always creating mock libwebrtc tracks
https://bugs.webkit.org/show_bug.cgi?id=169658
Patch by Youenn Fablet <youenn@apple.com> on 2017-03-15
Reviewed by Alex Christensen.
Source/WebCore:
Tests: webrtc/peer-connection-audio-mute.html
webrtc/video-mute.html
Creating real libwebrtc av tracks in case of RealTwoPeerConnections mock factory.
* testing/MockLibWebRTCPeerConnection.cpp:
(WebCore::MockLibWebRTCPeerConnectionFactory::CreateVideoTrack):
(WebCore::MockLibWebRTCPeerConnectionFactory::CreateAudioTrack):
* testing/MockLibWebRTCPeerConnection.h:
LayoutTests:
* TestExpectations:
* webrtc/audio-peer-connection-webaudio.html:
* webrtc/peer-connection-audio-mute-expected.txt: Added.
* webrtc/peer-connection-audio-mute.html: Added.
* webrtc/routines.js:
(analyseAudio):
* webrtc/video-expected.txt:
* webrtc/video-mute-expected.txt: Added.
* webrtc/video-mute.html: Added.
* webrtc/video.html:</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkLayoutTestsChangeLog">trunk/LayoutTests/ChangeLog</a></li>
<li><a href="#trunkLayoutTestsTestExpectations">trunk/LayoutTests/TestExpectations</a></li>
<li><a href="#trunkLayoutTestsTestExpectationsorig">trunk/LayoutTests/TestExpectations.orig</a></li>
<li><a href="#trunkLayoutTestswebrtcaudiopeerconnectionwebaudiohtml">trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html</a></li>
<li><a href="#trunkLayoutTestswebrtcroutinesjs">trunk/LayoutTests/webrtc/routines.js</a></li>
<li><a href="#trunkLayoutTestswebrtcvideoexpectedtxt">trunk/LayoutTests/webrtc/video-expected.txt</a></li>
<li><a href="#trunkLayoutTestswebrtcvideohtml">trunk/LayoutTests/webrtc/video.html</a></li>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp</a></li>
<li><a href="#trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh">trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h</a></li>
</ul>
<h3>Added Paths</h3>
<ul>
<li><a href="#trunkLayoutTestswebrtcpeerconnectionaudiomuteexpectedtxt">trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt</a></li>
<li><a href="#trunkLayoutTestswebrtcpeerconnectionaudiomutehtml">trunk/LayoutTests/webrtc/peer-connection-audio-mute.html</a></li>
<li><a href="#trunkLayoutTestswebrtcvideomuteexpectedtxt">trunk/LayoutTests/webrtc/video-mute-expected.txt</a></li>
<li><a href="#trunkLayoutTestswebrtcvideomutehtml">trunk/LayoutTests/webrtc/video-mute.html</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkLayoutTestsChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/ChangeLog (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/ChangeLog 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/ChangeLog 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -1,5 +1,23 @@
</span><span class="cx"> 2017-03-15 Youenn Fablet <youenn@apple.com>
</span><span class="cx">
</span><ins>+ run-webkit-tests is always creating mock libwebrtc tracks
+ https://bugs.webkit.org/show_bug.cgi?id=169658
+
+ Reviewed by Alex Christensen.
+
+ * TestExpectations:
+ * webrtc/audio-peer-connection-webaudio.html:
+ * webrtc/peer-connection-audio-mute-expected.txt: Added.
+ * webrtc/peer-connection-audio-mute.html: Added.
+ * webrtc/routines.js:
+ (analyseAudio):
+ * webrtc/video-expected.txt:
+ * webrtc/video-mute-expected.txt: Added.
+ * webrtc/video-mute.html: Added.
+ * webrtc/video.html:
+
+2017-03-15 Youenn Fablet <youenn@apple.com>
+
</ins><span class="cx"> Preventive clean-up: ensure RTCPeerConnection stays valid when calling postTask
</span><span class="cx"> https://bugs.webkit.org/show_bug.cgi?id=169661
</span><span class="cx">
</span></span></pre></div>
<a id="trunkLayoutTestsTestExpectations"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/TestExpectations (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/TestExpectations 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/TestExpectations 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -708,7 +708,7 @@
</span><span class="cx"> media/session [ Skip ]
</span><span class="cx">
</span><span class="cx"> # WebRTC backend not enabled by default on Mac/iOS release bots.
</span><del>-# GTK enables some of this tests on their TestExpectations file.
</del><ins>+# GTK enables some of these tests on their TestExpectations file.
</ins><span class="cx"> [ Release ] webrtc [ Skip ]
</span><span class="cx">
</span><span class="cx"> [ Debug ] webrtc/audio-peer-connection-webaudio.html [ Failure ]
</span></span></pre></div>
<a id="trunkLayoutTestsTestExpectationsorig"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/TestExpectations.orig (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/TestExpectations.orig 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/TestExpectations.orig 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -733,6 +733,18 @@
</span><span class="cx"> imported/w3c/web-platform-tests/XMLHttpRequest/open-url-redirected-worker-origin.htm [ Skip ]
</span><span class="cx"> imported/w3c/web-platform-tests/html/webappapis/system-state-and-capabilities/the-navigator-object/NavigatorID.html [ Skip ]
</span><span class="cx"> imported/w3c/web-platform-tests/html/webappapis/system-state-and-capabilities/the-navigator-object/NavigatorID.worker.html [ Skip ]
</span><ins>+imported/w3c/web-platform-tests/XMLHttpRequest/anonymous-mode-unsupported.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/open-after-setrequestheader.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/open-referer.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/send-accept-language.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-allow-empty-value.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-allow-whitespace-in-value.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-case-insensitive.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-header-allowed.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-header-forbidden.htm [ Failure ]
+imported/w3c/web-platform-tests/XMLHttpRequest/setrequestheader-open-setrequestheader.htm [ Failure ]
+imported/w3c/web-platform-tests/html/dom/interfaces.worker.html [ Failure ]
+imported/w3c/web-platform-tests/html/webappapis/scripting/events/event-handler-attributes-body-window.html [ Failure ]
</ins><span class="cx">
</span><span class="cx"> # Only iOS WK1 has testRunner.setPagePaused.
</span><span class="cx"> fast/dom/timer-fire-after-page-pause.html [ Skip ]
</span></span></pre></div>
<a id="trunkLayoutTestswebrtcaudiopeerconnectionwebaudiohtml"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/webrtc/audio-peer-connection-webaudio.html 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -7,75 +7,28 @@
</span><span class="cx"> <script src="../resources/testharnessreport.js"></script>
</span><span class="cx"> <script src ="routines.js"></script>
</span><span class="cx"> <script>
</span><del>- var test = async_test(() => {
</del><ins>+ promise_test((test) => {
</ins><span class="cx"> if (window.testRunner)
</span><span class="cx"> testRunner.setUserMediaPermission(true);
</span><span class="cx">
</span><del>- var heardHum = false;
- var heardBop = false;
- var heardBip = false;
-
- navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
</del><ins>+ return navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
</ins><span class="cx"> if (window.internals)
</span><span class="cx"> internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
</span><del>-
- createConnections((firstConnection) => {
- firstConnection.addStream(stream);
- }, (secondConnection) => {
- secondConnection.onaddstream = (streamEvent) => {
- var context = new webkitAudioContext();
- var sourceNode = context.createMediaStreamSource(streamEvent.stream);
- var analyser = context.createAnalyser();
- var gain = context.createGain();
-
- analyser.fftSize = 2048;
- analyser.smoothingTimeConstant = 0;
- analyser.minDecibels = -100;
- analyser.maxDecibels = 0;
- gain.gain.value = 0;
-
- sourceNode.connect(analyser);
- analyser.connect(gain);
- gain.connect(context.destination);
-
- function analyse() {
- var freqDomain = new Uint8Array(analyser.frequencyBinCount);
- analyser.getByteFrequencyData(freqDomain);
-
- var hasFrequency = expectedFrequency => {
- var bin = Math.floor(expectedFrequency * analyser.fftSize / context.sampleRate);
- return bin < freqDomain.length && freqDomain[bin] >= 150;
- };
-
- if (!heardHum)
- heardHum = hasFrequency(150);
-
- if (!heardBip)
- heardBip = hasFrequency(1500);
-
- if (!heardBop)
- heardBop = hasFrequency(500);
-
- if (heardHum && heardBip && heardBop)
- done();
- };
-
- var done = test.step_func_done(() => {
- clearTimeout(timeout);
- clearInterval(interval);
-
- assert_true(heardHum, "heard hum");
- assert_true(heardBip, "heard bip");
- assert_true(heardBop, "heard bop");
- test.done();
- });
-
- var timeout = setTimeout(done, 3000);
- var interval = setInterval(analyse, 1000 / 30);
- analyse();
- }
</del><ins>+ return new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ firstConnection.addStream(stream);
+ }, (secondConnection) => {
+ secondConnection.onaddstream = (streamEvent) => { resolve(streamEvent.stream); };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ }).then((stream) => {
+ return analyseAudio(stream, 1000);
+ }).then((results) => {
+ assert_true(results.heardHum, "heard hum");
+ assert_true(results.heardBip, "heard bip");
+ assert_true(results.heardBop, "heard bop");
</ins><span class="cx"> });
</span><del>- });
</del><ins>+ });
</ins><span class="cx"> }, "Basic audio playback through a peer connection");
</span><span class="cx"> </script>
</span><span class="cx"> </head>
</span></span></pre></div>
<a id="trunkLayoutTestswebrtcpeerconnectionaudiomuteexpectedtxt"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt (0 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute-expected.txt 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -0,0 +1,3 @@
</span><ins>+
+FAIL Muting and unmuting an audio track assert_true: heard hum expected true got false
+
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtcpeerconnectionaudiomutehtml"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/peer-connection-audio-mute.html (0 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/peer-connection-audio-mute.html (rev 0)
+++ trunk/LayoutTests/webrtc/peer-connection-audio-mute.html 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -0,0 +1,65 @@
</span><ins>+<!DOCTYPE html>
+<html>
+<head>
+ <meta charset="utf-8">
+ <title>Testing local audio capture playback causes "playing" event to fire</title>
+ <script src="../resources/testharness.js"></script>
+ <script src="../resources/testharnessreport.js"></script>
+</head>
+<body>
+ <script src ="routines.js"></script>
+ <script>
+ promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({audio: true}).then((stream) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ var stream;
+ return new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ firstConnection.addStream(stream);
+ }, (secondConnection) => {
+ secondConnection.onaddstream = (streamEvent) => {
+ stream = streamEvent.stream;
+ resolve();
+ };
+ });
+ }).then(() => {
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(stream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum");
+ assert_true(results.heardBip, "heard bip");
+ assert_true(results.heardBop, "heard bop");
+ });
+ }).then(() => {
+ stream.getAudioTracks().forEach((track) => {
+ track.enabled = false;
+ });
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(stream, 500).then((results) => {
+ assert_false(results.heardHum, "heard hum");
+ assert_false(results.heardBip, "heard bip");
+ assert_false(results.heardBop, "heard bop");
+ });
+ }).then(() => {
+ stream.getAudioTracks().forEach((track) => {
+ track.enabled = true;
+ });
+ return waitFor(500);
+ }).then(() => {
+ return analyseAudio(stream, 500).then((results) => {
+ assert_true(results.heardHum, "heard hum");
+ assert_true(results.heardBip, "heard bip");
+ assert_true(results.heardBop, "heard bop");
+ });
+ });
+ });
+ }, "Muting and unmuting an audio track");
+ </script>
+</body>
+</html>
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtcroutinesjs"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/webrtc/routines.js (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/routines.js 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/webrtc/routines.js 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -71,3 +71,62 @@
</span><span class="cx"> {
</span><span class="cx"> assert_unreached();
</span><span class="cx"> }
</span><ins>+
+function analyseAudio(stream, duration)
+{
+ return new Promise((resolve, reject) => {
+ var context = new webkitAudioContext();
+ var sourceNode = context.createMediaStreamSource(stream);
+ var analyser = context.createAnalyser();
+ var gain = context.createGain();
+
+ var results = { heardHum: false, heardBip: false, heardBop: false };
+
+ analyser.fftSize = 2048;
+ analyser.smoothingTimeConstant = 0;
+ analyser.minDecibels = -100;
+ analyser.maxDecibels = 0;
+ gain.gain.value = 0;
+
+ sourceNode.connect(analyser);
+ analyser.connect(gain);
+ gain.connect(context.destination);
+
+ function analyse() {
+ var freqDomain = new Uint8Array(analyser.frequencyBinCount);
+ analyser.getByteFrequencyData(freqDomain);
+
+ var hasFrequency = expectedFrequency => {
+ var bin = Math.floor(expectedFrequency * analyser.fftSize / context.sampleRate);
+ return bin < freqDomain.length && freqDomain[bin] >= 150;
+ };
+
+ if (!results.heardHum)
+ results.heardHum = hasFrequency(150);
+
+ if (!results.heardBip)
+ results.heardBip = hasFrequency(1500);
+
+ if (!results.heardBop)
+ results.heardBop = hasFrequency(500);
+
+ if (results.heardHum && results.heardBip && results.heardBop)
+ done();
+ };
+
+ function done() {
+ clearTimeout(timeout);
+ clearInterval(interval);
+ resolve(results);
+ }
+
+ var timeout = setTimeout(done, 3 * duration);
+ var interval = setInterval(analyse, duration / 30);
+ analyse();
+ });
+}
+
+function waitFor(duration)
+{
+ return new Promise((resolve) => setTimeout(resolve, duration));
+}
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtcvideoexpectedtxt"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/webrtc/video-expected.txt (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/video-expected.txt 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/webrtc/video-expected.txt 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -1,4 +1,4 @@
</span><span class="cx">
</span><span class="cx">
</span><del>-FAIL Basic video exchange assert_true: expected true got false
</del><ins>+PASS Basic video exchange
</ins><span class="cx">
</span></span></pre></div>
<a id="trunkLayoutTestswebrtcvideomuteexpectedtxt"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/video-mute-expected.txt (0 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/video-mute-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/video-mute-expected.txt 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -0,0 +1,4 @@
</span><ins>+
+
+PASS Video muted/unmuted track
+
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtcvideomutehtml"></a>
<div class="addfile"><h4>Added: trunk/LayoutTests/webrtc/video-mute.html (0 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/video-mute.html (rev 0)
+++ trunk/LayoutTests/webrtc/video-mute.html 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -0,0 +1,69 @@
</span><ins>+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing basic video exchange from offerer to receiver</title>
+ <script src="../resources/testharness.js"></script>
+ <script src="../resources/testharnessreport.js"></script>
+ </head>
+ <body>
+ <video id="video" autoplay=""></video>
+ <canvas id="canvas" width="640" height="480"></canvas>
+ <script src ="routines.js"></script>
+ <script>
+video = document.getElementById("video");
+canvas = document.getElementById("canvas");
+// FIXME: We should use tracks
+
+function isVideoBlack()
+{
+ canvas.width = video.videoWidth;
+ canvas.height = video.videoHeight;
+ canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height);
+
+ imageData = canvas.getContext('2d').getImageData(10, 325, 250, 1);
+ data = imageData.data;
+ for (var cptr = 0; cptr < canvas.width * canvas.height; ++cptr) {
+ if (data[4 * cptr] || data[4 * cptr + 1] || data[4 * cptr + 2])
+ return false;
+ }
+ return true;
+}
+
+var track;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ return navigator.mediaDevices.getUserMedia({ video: true}).then((stream) => {
+ return new Promise((resolve, reject) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ createConnections((firstConnection) => {
+ firstConnection.addStream(stream);
+ }, (secondConnection) => {
+ secondConnection.onaddstream = (streamEvent) => { resolve(streamEvent.stream); };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then((stream) => {
+ video.srcObject = stream;
+ track = stream.getVideoTracks()[0];
+ return video.play();
+ }).then(() => {
+ assert_false(isVideoBlack());
+ }).then(() => {
+ track.enabled = false;
+ return waitFor(500);
+ }).then(() => {
+ assert_true(isVideoBlack());
+ track.enabled = true;
+ return waitFor(500);
+ }).then(() => {
+ assert_false(isVideoBlack());
+ });
+}, "Video muted/unmuted track");
+ </script>
+ </body>
+</html>
</ins></span></pre></div>
<a id="trunkLayoutTestswebrtcvideohtml"></a>
<div class="modfile"><h4>Modified: trunk/LayoutTests/webrtc/video.html (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/LayoutTests/webrtc/video.html 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/LayoutTests/webrtc/video.html 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -11,12 +11,6 @@
</span><span class="cx"> <canvas id="canvas" width="640" height="480"></canvas>
</span><span class="cx"> <script src ="routines.js"></script>
</span><span class="cx"> <script>
</span><del>-if (window.internals)
- internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
-
-if (window.testRunner)
- testRunner.setUserMediaPermission(true);
-
</del><span class="cx"> video = document.getElementById("video");
</span><span class="cx"> canvas = document.getElementById("canvas");
</span><span class="cx"> // FIXME: We should use tracks
</span><span class="lines">@@ -23,56 +17,50 @@
</span><span class="cx">
</span><span class="cx"> function testImage()
</span><span class="cx"> {
</span><del>- try {
- canvas.width = video.videoWidth;
- canvas.height = video.videoHeight;
- canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height);
</del><ins>+ canvas.width = video.videoWidth;
+ canvas.height = video.videoHeight;
+ canvas.getContext('2d').drawImage(video, 0, 0, canvas.width, canvas.height);
</ins><span class="cx">
</span><del>- imageData = canvas.getContext('2d').getImageData(10, 325, 250, 1);
- data = imageData.data;
</del><ins>+ imageData = canvas.getContext('2d').getImageData(10, 325, 250, 1);
+ data = imageData.data;
</ins><span class="cx">
</span><del>- var index = 20;
- assert_true(data[index] < 100);
- assert_true(data[index + 1] < 100);
- assert_true(data[index + 2] < 100);
</del><ins>+ var index = 20;
+ assert_true(data[index] < 100);
+ assert_true(data[index + 1] < 100);
+ assert_true(data[index + 2] < 100);
</ins><span class="cx">
</span><del>- index = 80;
- assert_true(data[index] > 200);
- assert_true(data[index + 1] > 200);
- assert_true(data[index + 2] > 200);
</del><ins>+ index = 80;
+ assert_true(data[index] > 200);
+ assert_true(data[index + 1] > 200);
+ assert_true(data[index + 2] > 200);
</ins><span class="cx">
</span><del>- index += 80;
- assert_true(data[index] > 200);
- assert_true(data[index + 1] > 200);
- assert_true(data[index + 2] < 100);
-
- finishTest();
- } catch(e) {
- errorTest(e);
- }
</del><ins>+ index += 80;
+ assert_true(data[index] > 200);
+ assert_true(data[index + 1] > 200);
+ assert_true(data[index + 2] < 100);
</ins><span class="cx"> }
</span><span class="cx">
</span><del>-function testStream(stream)
-{
- video.srcObject = stream;
- // Video may play with black frames
- video.onplay = setTimeout(() => {
- testImage();
- }, 1000);
-}
</del><ins>+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
</ins><span class="cx">
</span><del>-var finishTest, errorTest;
-promise_test((test) => {
</del><span class="cx"> return navigator.mediaDevices.getUserMedia({ video: true}).then((stream) => {
</span><span class="cx"> return new Promise((resolve, reject) => {
</span><del>- finishTest = resolve;
- errorTest = reject;
</del><ins>+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
</ins><span class="cx"> createConnections((firstConnection) => {
</span><span class="cx"> firstConnection.addStream(stream);
</span><span class="cx"> }, (secondConnection) => {
</span><del>- secondConnection.onaddstream = (streamEvent) => { testStream(streamEvent.stream); };
</del><ins>+ secondConnection.onaddstream = (streamEvent) => { resolve(streamEvent.stream); };
</ins><span class="cx"> });
</span><ins>+ setTimeout(() => reject("Test timed out"), 5000);
</ins><span class="cx"> });
</span><ins>+ }).then((stream) => {
+ video.srcObject = stream;
+ return video.play();
+ }).then(() => {
+ testImage();
</ins><span class="cx"> });
</span><span class="cx"> }, "Basic video exchange");
</span><span class="cx"> </script>
</span></span></pre></div>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/Source/WebCore/ChangeLog 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -1,5 +1,22 @@
</span><span class="cx"> 2017-03-15 Youenn Fablet <youenn@apple.com>
</span><span class="cx">
</span><ins>+ run-webkit-tests is always creating mock libwebrtc tracks
+ https://bugs.webkit.org/show_bug.cgi?id=169658
+
+ Reviewed by Alex Christensen.
+
+ Tests: webrtc/peer-connection-audio-mute.html
+ webrtc/video-mute.html
+
+ Creating real libwebrtc av tracks in case of RealTwoPeerConnections mock factory.
+
+ * testing/MockLibWebRTCPeerConnection.cpp:
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreateVideoTrack):
+ (WebCore::MockLibWebRTCPeerConnectionFactory::CreateAudioTrack):
+ * testing/MockLibWebRTCPeerConnection.h:
+
+2017-03-15 Youenn Fablet <youenn@apple.com>
+
</ins><span class="cx"> Preventive clean-up: ensure RTCPeerConnection stays valid when calling postTask
</span><span class="cx"> https://bugs.webkit.org/show_bug.cgi?id=169661
</span><span class="cx">
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectioncpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -189,6 +189,20 @@
</span><span class="cx"> return new rtc::RefCountedObject<MockLibWebRTCPeerConnection>(*observer);
</span><span class="cx"> }
</span><span class="cx">
</span><ins>+rtc::scoped_refptr<webrtc::VideoTrackInterface> MockLibWebRTCPeerConnectionFactory::CreateVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source)
+{
+ if (m_testCase == "TwoRealPeerConnections")
+ return realPeerConnectionFactory()->CreateVideoTrack(id, source);
+ return new rtc::RefCountedObject<MockLibWebRTCVideoTrack>(id, source);
+}
+
+rtc::scoped_refptr<webrtc::AudioTrackInterface> MockLibWebRTCPeerConnectionFactory::CreateAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source)
+{
+ if (m_testCase == "TwoRealPeerConnections")
+ return realPeerConnectionFactory()->CreateAudioTrack(id, source);
+ return new rtc::RefCountedObject<MockLibWebRTCAudioTrack>(id, source);
+}
+
</ins><span class="cx"> rtc::scoped_refptr<webrtc::MediaStreamInterface> MockLibWebRTCPeerConnectionFactory::CreateLocalMediaStream(const std::string& label)
</span><span class="cx"> {
</span><span class="cx"> return new rtc::RefCountedObject<webrtc::MediaStream>(label);
</span></span></pre></div>
<a id="trunkSourceWebCoretestingMockLibWebRTCPeerConnectionh"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h (213982 => 213983)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-03-15 16:35:18 UTC (rev 213982)
+++ trunk/Source/WebCore/testing/MockLibWebRTCPeerConnection.h 2017-03-15 16:38:10 UTC (rev 213983)
</span><span class="lines">@@ -244,8 +244,9 @@
</span><span class="cx"> rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*) final { return nullptr; }
</span><span class="cx"> rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*, const webrtc::MediaConstraintsInterface*) final { return nullptr; }
</span><span class="cx">
</span><del>- rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateVideoTrack(const std::string& id, webrtc::VideoTrackSourceInterface* source) final { return new rtc::RefCountedObject<MockLibWebRTCVideoTrack>(id, source); }
- rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateAudioTrack(const std::string& id, webrtc::AudioSourceInterface* source) final { return new rtc::RefCountedObject<MockLibWebRTCAudioTrack>(id, source); }
</del><ins>+ rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateVideoTrack(const std::string&, webrtc::VideoTrackSourceInterface*) final;
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateAudioTrack(const std::string&, webrtc::AudioSourceInterface*) final;
+
</ins><span class="cx"> bool StartAecDump(rtc::PlatformFile, int64_t) final { return false; }
</span><span class="cx"> void StopAecDump() final { }
</span><span class="cx">
</span></span></pre>
</div>
</div>
</body>
</html>