Title: [214348] trunk
- Revision
- 214348
- Author
- [email protected]
- Date
- 2017-03-24 09:25:09 -0700 (Fri, 24 Mar 2017)
Log Message
Fix framesEncoded/framesDecoded RTC stats
https://bugs.webkit.org/show_bug.cgi?id=170024
Patch by Youenn Fablet <[email protected]> on 2017-03-24
Reviewed by Eric Carlson.
Source/WebCore:
Test: webrtc/video-stats.html
Adding access to these fields now that they are available.
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::fillInboundRTPStreamStats):
(WebCore::fillOutboundRTPStreamStats):
LayoutTests:
* webrtc/video-stats-expected.txt: Added.
* webrtc/video-stats.html: Added.
Modified Paths
Added Paths
Diff
Modified: trunk/LayoutTests/ChangeLog (214347 => 214348)
--- trunk/LayoutTests/ChangeLog 2017-03-24 14:55:59 UTC (rev 214347)
+++ trunk/LayoutTests/ChangeLog 2017-03-24 16:25:09 UTC (rev 214348)
@@ -1,3 +1,13 @@
+2017-03-24 Youenn Fablet <[email protected]>
+
+ Fix framesEncoded/framesDecoded RTC stats
+ https://bugs.webkit.org/show_bug.cgi?id=170024
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/video-stats-expected.txt: Added.
+ * webrtc/video-stats.html: Added.
+
2017-03-24 Carlos Garcia Campos <[email protected]>
Unreviewed GTK+ gardening. Mark media/video-load-require-user-gesture.html as failing after r214338.
Added: trunk/LayoutTests/webrtc/video-stats-expected.txt (0 => 214348)
--- trunk/LayoutTests/webrtc/video-stats-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/video-stats-expected.txt 2017-03-24 16:25:09 UTC (rev 214348)
@@ -0,0 +1,3 @@
+
+PASS Basic video stats
+
Added: trunk/LayoutTests/webrtc/video-stats.html (0 => 214348)
--- trunk/LayoutTests/webrtc/video-stats.html (rev 0)
+++ trunk/LayoutTests/webrtc/video-stats.html 2017-03-24 16:25:09 UTC (rev 214348)
@@ -0,0 +1,100 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing basic video exchange from offerer to receiver</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <script src =""
+ <script>
+
+function getInboundRTPStats(connection)
+{
+ return connection.getStats().then((report) => {
+ var stats;
+ report.forEach((statItem) => {
+ if (statItem.type === "inbound-rtp") {
+ stats = statItem;
+ }
+ });
+ return stats;
+ });
+}
+
+function getOutboundRTPStats(connection)
+{
+ return connection.getStats().then((report) => {
+ var stats;
+ report.forEach((statItem) => {
+ if (statItem.type === "outbound-rtp") {
+ stats = statItem;
+ }
+ });
+ return stats;
+ });
+}
+
+function testStatsTwice(firstConnection, secondConnection)
+{
+ return Promise.all([
+ firstConnection.getStats().then((report) => {
+ firstReportB = report;
+ }),
+ secondConnection.getStats().then((report) => {
+ secondReportB = report;
+ })
+ ]);
+}
+
+var firstConnection, secondConnection;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ var localStream, remoteStream;
+ return navigator.mediaDevices.getUserMedia({ video: true}).then((stream) => {
+ localStream = stream;
+ return new Promise((resolve, reject) => {
+ if (window.internals)
+ internals.useMockRTCPeerConnectionFactory("TwoRealPeerConnections");
+
+ createConnections((connection) => {
+ firstConnection = connection;
+ firstConnection.addTrack(stream.getVideoTracks()[0], stream);
+ }, (connection) => {
+ secondConnection = connection;
+ secondConnection._ontrack_ = (trackEvent) => {
+ remoteStream = trackEvent.streams[0];
+ resolve();
+ };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then(() => {
+ return getOutboundRTPStats(firstConnection);
+ }).then((stats) => {
+ assert_true(!!stats, "outbound-rtp stats should not be null");
+ assert_true(Number.isInteger(stats.framesEncoded), "framesEncoded should be an integer");
+ statsFirstConnection = stats;
+ return getInboundRTPStats(secondConnection);
+ }).then((stats) => {
+ assert_true(!!stats, "inbound-rtp stats should not be null");
+ assert_true(Number.isInteger(stats.framesDecoded), "framesDecoded should be an integer");
+ statsSecondConnection = stats;
+ return waitFor(300);
+ }).then(() => {
+ return getOutboundRTPStats(firstConnection);
+ }).then((stats) => {
+ assert_true(stats.timestamp > statsFirstConnection.timestamp, "Timestamp for first connection should have increased");
+ assert_true(stats.framesEncoded > statsFirstConnection.framesEncoded, "Number of encoded frames should have increased");
+ return getInboundRTPStats(secondConnection);
+ }).then((stats) => {
+ assert_true(stats.timestamp > statsSecondConnection.timestamp, "Timestamp for second connection should have increased");
+ assert_true(stats.framesDecoded > statsSecondConnection.framesDecoded, "Number of decoded frames should have increased");
+ });
+}, "Basic video stats");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/WebCore/ChangeLog (214347 => 214348)
--- trunk/Source/WebCore/ChangeLog 2017-03-24 14:55:59 UTC (rev 214347)
+++ trunk/Source/WebCore/ChangeLog 2017-03-24 16:25:09 UTC (rev 214348)
@@ -1,3 +1,18 @@
+2017-03-24 Youenn Fablet <[email protected]>
+
+ Fix framesEncoded/framesDecoded RTC stats
+ https://bugs.webkit.org/show_bug.cgi?id=170024
+
+ Reviewed by Eric Carlson.
+
+ Test: webrtc/video-stats.html
+
+ Adding access to these fields now that they are available.
+
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::fillInboundRTPStreamStats):
+ (WebCore::fillOutboundRTPStreamStats):
+
2017-03-24 Carlos Garcia Campos <[email protected]>
Unreviewed. Fix GTK+ test /webkit2/WebKitWebView/default-menu after r214244.
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (214347 => 214348)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-24 14:55:59 UTC (rev 214347)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-24 16:25:09 UTC (rev 214348)
@@ -305,8 +305,8 @@
stats.gapLossRate = *rtcStats.gap_loss_rate;
if (rtcStats.gap_discard_rate.is_defined())
stats.gapDiscardRate = *rtcStats.gap_discard_rate;
- // FIXME: Set framesDecoded
- stats.framesDecoded = 0;
+ if (rtcStats.frames_decoded.is_defined())
+ stats.framesDecoded = *rtcStats.frames_decoded;
}
static inline void fillOutboundRTPStreamStats(RTCStatsReport::OutboundRTPStreamStats& stats, const webrtc::RTCOutboundRTPStreamStats& rtcStats)
@@ -319,8 +319,8 @@
stats.bytesSent = *rtcStats.bytes_sent;
if (rtcStats.target_bitrate.is_defined())
stats.targetBitrate = *rtcStats.target_bitrate;
- // FIXME: Set framesEncoded
- stats.framesEncoded = 0;
+ if (rtcStats.frames_encoded.is_defined())
+ stats.framesEncoded = *rtcStats.frames_encoded;
}
void LibWebRTCMediaEndpoint::StatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& rtcReport)
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