Diff
Modified: trunk/LayoutTests/ChangeLog (236345 => 236346)
--- trunk/LayoutTests/ChangeLog 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/LayoutTests/ChangeLog 2018-09-21 19:39:41 UTC (rev 236346)
@@ -1,3 +1,13 @@
+2018-09-21 Youenn Fablet <[email protected]>
+
+ Add RTCCodecStats support
+ https://bugs.webkit.org/show_bug.cgi?id=189792
+ <rdar://problem/32370668>
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/video-stats.html:
+
2018-09-21 Megan Gardner <[email protected]>
[iOS] Layout test fast/gradients/conic-gradient-alpha.html is failing
Modified: trunk/LayoutTests/imported/w3c/ChangeLog (236345 => 236346)
--- trunk/LayoutTests/imported/w3c/ChangeLog 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/LayoutTests/imported/w3c/ChangeLog 2018-09-21 19:39:41 UTC (rev 236346)
@@ -1,3 +1,15 @@
+2018-09-21 Youenn Fablet <[email protected]>
+
+ Add RTCCodecStats support
+ https://bugs.webkit.org/show_bug.cgi?id=189792
+ <rdar://problem/32370668>
+
+ Reviewed by Eric Carlson.
+
+ * web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt:
+ * web-platform-tests/webrtc/RTCRtpReceiver-getStats.https-expected.txt:
+ * web-platform-tests/webrtc/RTCRtpSender-getStats.https-expected.txt:
+
2018-09-20 Frederic Wang <[email protected]>
Synchronize CSSOM View test suite against upstream WPT
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https-expected.txt (236345 => 236346)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https-expected.txt 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https-expected.txt 2018-09-21 19:39:41 UTC (rev 236346)
@@ -1,4 +1,4 @@
FAIL receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats assert_true: Expect statsReport to contain stats object of type inbound-rtp expected true got false
-FAIL receiver.getStats() via addTrack should return stats report containing inbound-rtp stats assert_equals: Expect dictionary.mediaType to be string expected "string" but got "undefined"
+FAIL receiver.getStats() via addTrack should return stats report containing inbound-rtp stats assert_true: Expect stats report to have stats object with id RTCTransport_0_1 expected true got false
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getStats.https-expected.txt (236345 => 236346)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getStats.https-expected.txt 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getStats.https-expected.txt 2018-09-21 19:39:41 UTC (rev 236346)
@@ -1,4 +1,4 @@
FAIL sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats assert_true: Expect statsReport to contain stats object of type outbound-rtp expected true got false
-FAIL sender.getStats() via addTrack should return stats report containing outbound-rtp stats assert_equals: Expect dictionary.trackId to be string expected "string" but got "undefined"
+FAIL sender.getStats() via addTrack should return stats report containing outbound-rtp stats assert_true: Expect stats report to have stats object with id RTCTransport_0_1 expected true got false
Modified: trunk/LayoutTests/webrtc/video-stats.html (236345 => 236346)
--- trunk/LayoutTests/webrtc/video-stats.html 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/LayoutTests/webrtc/video-stats.html 2018-09-21 19:39:41 UTC (rev 236346)
@@ -69,6 +69,8 @@
if (stats.framesDecoded > statsSecondConnection.framesDecoded) {
if (testTimestampDifference(stats.timestamp - statsSecondConnection.timestamp, stats.framesDecoded - statsSecondConnection.framesDecoded))
return Promise.reject("timestamp and frames increment do not match");
+ assert_not_equals(Object.keys(stats).indexOf("codecId"), -1, "codecId");
+ assert_not_equals(Object.keys(stats).indexOf("trackId"), -1, "trackId");
return;
}
if (++count === 20)
@@ -85,6 +87,8 @@
if (stats.framesEncoded > statsFirstConnection.framesEncoded) {
if (testTimestampDifference(stats.timestamp - statsFirstConnection.timestamp, stats.framesEncoded - statsFirstConnection.framesEncoded))
return Promise.reject("timestamp and frames increment do not match");
+ assert_not_equals(Object.keys(stats).indexOf("codecId"), -1, "codecId");
+ assert_not_equals(Object.keys(stats).indexOf("trackId"), -1, "trackId");
return;
}
if (++count === 20)
@@ -133,10 +137,10 @@
await checkOutboundFramesNumberIncreased(firstConnection, statsFirstConnection, 0);
let types = await getStatsType(firstConnection);
- assert_array_equals(types, ["candidate-pair", "certificate", "inbound-rtp", "outbound-rtp", "track"]);
+ assert_array_equals(types, ["candidate-pair", "certificate", "codec", "inbound-rtp", "outbound-rtp", "track"]);
types = await getStatsType(secondConnection);
- assert_array_equals(types, ["candidate-pair", "certificate", "inbound-rtp", "outbound-rtp", "track"]);
+ assert_array_equals(types, ["candidate-pair", "certificate", "codec", "inbound-rtp", "outbound-rtp", "track"]);
}, "Basic video stats");
promise_test(async (test) => {
Modified: trunk/Source/WebCore/ChangeLog (236345 => 236346)
--- trunk/Source/WebCore/ChangeLog 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/Source/WebCore/ChangeLog 2018-09-21 19:39:41 UTC (rev 236346)
@@ -1,3 +1,26 @@
+2018-09-21 Youenn Fablet <[email protected]>
+
+ Add RTCCodecStats support
+ https://bugs.webkit.org/show_bug.cgi?id=189792
+ <rdar://problem/32370668>
+
+ Reviewed by Eric Carlson.
+
+ Covered by updated and rebased tests.
+
+ * Modules/mediastream/RTCStatsReport.h:
+ Removed fields that are already defined in the base class.
+ (WebCore::RTCStatsReport::CodecStats::CodecStats):
+ Add support for RTCCodecStats.
+ * Modules/mediastream/RTCStatsReport.idl:
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::OnStatsDelivered):
+ * Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp:
+ (WebCore::fillRTCRTPStreamStats):
+ (WebCore::fillRTCCodecStats):
+ (WebCore::LibWebRTCStatsCollector::OnStatsDelivered):
+ Add routines to fill RTCCodecStats from libwebrtc stats.
+
2018-09-20 Simon Fraser <[email protected]>
Make "overflow: overlay" a synonym for "overflow: auto"
Modified: trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h (236345 => 236346)
--- trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h 2018-09-21 19:39:41 UTC (rev 236346)
@@ -64,7 +64,7 @@
bool isRemote { false };
String mediaType;
- String mediaTrackId;
+ String trackId;
String transportId;
String codecId;
unsigned long firCount { 0 };
@@ -77,18 +77,6 @@
struct InboundRTPStreamStats : RTCRTPStreamStats {
InboundRTPStreamStats() { type = RTCStatsReport::Type::InboundRtp; }
- uint32_t ssrc;
- String associateStatsId;
- bool isRemote { false };
- String mediaType;
- String mediaTrackId;
- String transportId;
- String codecId;
- unsigned long firCount { 0 };
- unsigned long pliCount { 0 };
- unsigned long nackCount { 0 };
- unsigned long sliCount { 0 };
- unsigned long long qpSum { 0 };
unsigned long packetsReceived { 0 };
unsigned long long bytesReceived { 0 };
unsigned long packetsLost { 0 };
@@ -198,6 +186,24 @@
String issuerCertificateId;
};
+ enum class CodecType {
+ Encode,
+ Decode
+ };
+
+ struct CodecStats : Stats {
+ CodecStats() { type = RTCStatsReport::Type::Codec; }
+
+ unsigned long payloadType;
+ std::optional<CodecType> codecType;
+ String transportId;
+ String mimeType;
+ std::optional<unsigned> clockRate;
+ std::optional<unsigned> channels;
+ String sdpFmtpLine;
+ String implementation;
+ };
+
private:
RTCStatsReport() = default;
Modified: trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.idl (236345 => 236346)
--- trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.idl 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.idl 2018-09-21 19:39:41 UTC (rev 236346)
@@ -59,7 +59,7 @@
DOMString associateStatsId;
boolean isRemote = false;
DOMString mediaType;
- DOMString mediaTrackId;
+ DOMString trackId;
DOMString transportId;
DOMString codecId;
unsigned long firCount;
@@ -173,7 +173,23 @@
DOMString issuerCertificateId;
};
-// FIXME 169662: missing RTCCodecStats
+enum RTCCodecType {
+ "encode",
+ "decode"
+};
+
+[ JSGenerateToJSObject ]
+dictionary RTCCodecStats : RTCStats {
+ unsigned long payloadType;
+ RTCCodecType codecType;
+ DOMString transportId;
+ DOMString mimeType;
+ unsigned long clockRate;
+ unsigned long channels;
+ DOMString sdpFmtpLine;
+ DOMString implementation;
+};
+
// FIXME 169662: missing RTCPeerConnectionStats
// FIXME 169662: missing RTCMediaStreamStats
// FIXME 169662: missing RTCTransportStats
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (236345 => 236346)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2018-09-21 19:39:41 UTC (rev 236346)
@@ -798,12 +798,8 @@
m_statsLogTimer.startRepeating(statsLogInterval(timestamp));
}
- for (auto iterator = report->begin(); iterator != report->end(); ++iterator) {
- if (iterator->type() == webrtc::RTCCodecStats::kType)
- continue;
-
+ for (auto iterator = report->begin(); iterator != report->end(); ++iterator)
ALWAYS_LOG(Logger::LogSiteIdentifier("LibWebRTCMediaEndpoint", "OnStatsDelivered", logIdentifier()), RTCStatsLogger { *iterator });
- }
});
#else
UNUSED_PARAM(report);
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp (236345 => 236346)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp 2018-09-21 19:30:20 UTC (rev 236345)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp 2018-09-21 19:39:41 UTC (rev 236346)
@@ -72,7 +72,7 @@
if (rtcStats.media_type.is_defined())
stats.mediaType = fromStdString(*rtcStats.media_type);
if (rtcStats.track_id.is_defined())
- stats.mediaTrackId = fromStdString(*rtcStats.track_id);
+ stats.trackId = fromStdString(*rtcStats.track_id);
if (rtcStats.transport_id.is_defined())
stats.transportId = fromStdString(*rtcStats.transport_id);
if (rtcStats.codec_id.is_defined())
@@ -291,6 +291,25 @@
if (rtcStats.issuer_certificate_id.is_defined())
stats.issuerCertificateId = fromStdString(*rtcStats.issuer_certificate_id);
}
+
+static inline void fillRTCCodecStats(RTCStatsReport::CodecStats& stats, const webrtc::RTCCodecStats& rtcStats)
+{
+ fillRTCStats(stats, rtcStats);
+
+ if (rtcStats.payload_type.is_defined())
+ stats.payloadType = *rtcStats.payload_type;
+ if (rtcStats.mime_type.is_defined())
+ stats.mimeType = fromStdString(*rtcStats.mime_type);
+ if (rtcStats.clock_rate.is_defined())
+ stats.clockRate = *rtcStats.clock_rate;
+ if (rtcStats.channels.is_defined())
+ stats.channels = *rtcStats.channels;
+ if (rtcStats.sdp_fmtp_line.is_defined())
+ stats.sdpFmtpLine = fromStdString(*rtcStats.sdp_fmtp_line);
+ if (rtcStats.implementation.is_defined())
+ stats.implementation = fromStdString(*rtcStats.implementation);
+}
+
void LibWebRTCStatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& rtcReport)
{
callOnMainThread([protectedThis = rtc::scoped_refptr<LibWebRTCStatsCollector>(this), rtcReport] {
@@ -325,6 +344,10 @@
RTCStatsReport::CertificateStats stats;
fillRTCCertificateStats(stats, static_cast<const webrtc::RTCCertificateStats&>(rtcStats));
report->addStats<IDLDictionary<RTCStatsReport::CertificateStats>>(WTFMove(stats));
+ } else if (rtcStats.type() == webrtc::RTCCodecStats::kType) {
+ RTCStatsReport::CodecStats stats;
+ fillRTCCodecStats(stats, static_cast<const webrtc::RTCCodecStats&>(rtcStats));
+ report->addStats<IDLDictionary<RTCStatsReport::CodecStats>>(WTFMove(stats));
}
}
});
