Diff
Modified: trunk/LayoutTests/ChangeLog (214419 => 214420)
--- trunk/LayoutTests/ChangeLog 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/LayoutTests/ChangeLog 2017-03-27 18:10:45 UTC (rev 214420)
@@ -1,3 +1,13 @@
+2017-03-27 Youenn Fablet <you...@apple.com>
+
+ Add support for RTCRtpReceiver/RTCRtpSender getParameters
+ https://bugs.webkit.org/show_bug.cgi?id=170057
+
+ Reviewed by Alex Christensen.
+
+ * webrtc/video-getParameters-expected.txt: Added.
+ * webrtc/video-getParameters.html: Added.
+
2017-03-27 Myles C. Maxfield <mmaxfi...@apple.com>
font variation properties don't need to accept numbers
Added: trunk/LayoutTests/webrtc/video-getParameters-expected.txt (0 => 214420)
--- trunk/LayoutTests/webrtc/video-getParameters-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/video-getParameters-expected.txt 2017-03-27 18:10:45 UTC (rev 214420)
@@ -0,0 +1,3 @@
+
+PASS Basic video stats
+
Added: trunk/LayoutTests/webrtc/video-getParameters.html (0 => 214420)
--- trunk/LayoutTests/webrtc/video-getParameters.html (rev 0)
+++ trunk/LayoutTests/webrtc/video-getParameters.html 2017-03-27 18:10:45 UTC (rev 214420)
@@ -0,0 +1,56 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing getParameters on a basic video exchange</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <script src =""
+ <script>
+var firstConnection, secondConnection;
+promise_test((test) => {
+ if (window.testRunner)
+ testRunner.setUserMediaPermission(true);
+
+ var sender, receiver;
+ 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;
+ sender = firstConnection.addTrack(stream.getVideoTracks()[0], stream);
+ }, (connection) => {
+ secondConnection = connection;
+ receiver = secondConnection.addTransceiver("video").receiver;
+ secondConnection._ontrack_ = (trackEvent) => {
+ remoteStream = trackEvent.streams[0];
+ resolve();
+ };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ }).then(() => {
+ assert_true(!!sender, "sender should not be undefined");
+ assert_true(!!receiver, "receiver should not be undefined");
+ var senderParameters = sender.getParameters();
+ var receiverParameters = receiver.getParameters();
+ assert_true(!!senderParameters, "sender parameters should not be undefined or null");
+ assert_true(!!receiverParameters, "receiver parameters should not be undefined or null");
+
+ assert_equals(JSON.stringify(senderParameters), JSON.stringify(receiverParameters), "testing sender vs. receiver parameters");
+
+ senderParameters.encodings[0].fec.ssrc = 1;
+ senderParameters.encodings[0].rtx.ssrc = 1;
+ senderParameters.encodings[0].ssrc = 1;
+ assert_equals(JSON.stringify(senderParameters), '{"codecs":[],"degradationPreference":"balanced","encodings":[{"active":true,"dtx":"disabled","fec":{"ssrc":1},"maxBitrate":0,"maxFramerate":0,"priority":"medium","rid":"","rtx":{"ssrc":1},"scaleResolutionDownBy":1,"ssrc":1}],"headerExtensions":[],"transactionId":""}', "Testing sanitized parameters");
+ });
+}, "Basic video stats");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (214419 => 214420)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2017-03-27 18:10:45 UTC (rev 214420)
@@ -1,3 +1,12 @@
+2017-03-27 Youenn Fablet <you...@apple.com>
+
+ Add support for RTCRtpReceiver/RTCRtpSender getParameters
+ https://bugs.webkit.org/show_bug.cgi?id=170057
+
+ Reviewed by Alex Christensen.
+
+ * Source/webrtc/api/mediatypes.h:
+
2017-03-22 Alex Christensen <achristen...@webkit.org>
Fix warnings in libwebrtc
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h (214419 => 214420)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -11,6 +11,8 @@
#ifndef WEBRTC_API_MEDIATYPES_H_
#define WEBRTC_API_MEDIATYPES_H_
+#include "webrtc/base/export.h"
+
#include <string>
namespace cricket {
@@ -21,7 +23,7 @@
MEDIA_TYPE_DATA
};
-std::string MediaTypeToString(MediaType type);
+WEBRTC_DYLIB_EXPORT std::string MediaTypeToString(MediaType type);
// Aborts on invalid string. Only expected to be used on strings that are
// guaranteed to be valid, such as MediaStreamTrackInterface::kind().
MediaType MediaTypeFromString(const std::string& type_str);
Modified: trunk/Source/WebCore/CMakeLists.txt (214419 => 214420)
--- trunk/Source/WebCore/CMakeLists.txt 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/CMakeLists.txt 2017-03-27 18:10:45 UTC (rev 214420)
@@ -250,6 +250,7 @@
Modules/mediastream/RTCPeerConnection.idl
Modules/mediastream/RTCPeerConnectionIceEvent.idl
Modules/mediastream/RTCPeerConnectionState.idl
+ Modules/mediastream/RTCRtpParameters.idl
Modules/mediastream/RTCRtpReceiver.idl
Modules/mediastream/RTCRtpSender.idl
Modules/mediastream/RTCRtpTransceiver.idl
Modified: trunk/Source/WebCore/ChangeLog (214419 => 214420)
--- trunk/Source/WebCore/ChangeLog 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/ChangeLog 2017-03-27 18:10:45 UTC (rev 214420)
@@ -1,3 +1,72 @@
+2017-03-27 Youenn Fablet <you...@apple.com>
+
+ Add support for RTCRtpReceiver/RTCRtpSender getParameters
+ https://bugs.webkit.org/show_bug.cgi?id=170057
+
+ Reviewed by Alex Christensen.
+
+ Test: webrtc/video-getParameters.html
+
+ getParameters returns a dictionary of values taken from libwebrtc RtpReceiverInterface/RtpSenderInrterface objects.
+ Added a direct link between WebCore RTCRtpReceiver and libwebrtc RtpReceiverInterface object.
+ Making the link between WebCore RTCRtpSender and libwebrtc RtpSenderInrterface object through
+ PeerConnectionBackend to keep the current architecture shared with OpenWebRTC.
+ In the future, we should try to make the link more direct.
+
+ Added routines to support the conversion from libwebrtc to WebCore.
+ Ensured that RTCRtpReceiver is cleaning its backend when being stopped.
+
+ * CMakeLists.txt:
+ * DerivedSources.make:
+ * Modules/mediastream/MediaEndpointPeerConnection.cpp:
+ (WebCore::MediaEndpointPeerConnection::setRemoteDescriptionTask):
+ * Modules/mediastream/PeerConnectionBackend.h:
+ (WebCore::PeerConnectionBackend::getParameters):
+ * Modules/mediastream/RTCPeerConnection.cpp:
+ (WebCore::RTCPeerConnection::doClose):
+ (WebCore::RTCPeerConnection::getParameters):
+ * Modules/mediastream/RTCPeerConnection.h:
+ * Modules/mediastream/RTCRtpParameters.h: Added.
+ * Modules/mediastream/RTCRtpParameters.idl: Added.
+ * Modules/mediastream/RTCRtpReceiver.cpp:
+ (WebCore::RTCRtpReceiver::RTCRtpReceiver):
+ * Modules/mediastream/RTCRtpReceiver.h:
+ (WebCore::RTCRtpReceiver::Backend::~Backend):
+ (WebCore::RTCRtpReceiver::Backend::getParameters):
+ (WebCore::RTCRtpReceiver::create):
+ (WebCore::RTCRtpReceiver::stop):
+ (WebCore::RTCRtpReceiver::setBackend):
+ (WebCore::RTCRtpReceiver::getParameters):
+ * Modules/mediastream/RTCRtpReceiver.idl:
+ * Modules/mediastream/RTCRtpSender.cpp:
+ (WebCore::RTCRtpSender::create):
+ (WebCore::RTCRtpSender::RTCRtpSender):
+ (WebCore::RTCRtpSender::replaceTrack):
+ (WebCore::RTCRtpSender::getParameters):
+ * Modules/mediastream/RTCRtpSender.h:
+ (WebCore::RTCRtpSender::Backend::~Backend):
+ (WebCore::RTCRtpSender::isStopped):
+ (WebCore::RTCRtpSender::stop):
+ * Modules/mediastream/RTCRtpSender.idl:
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::addTrack):
+ (WebCore::LibWebRTCMediaEndpoint::addRemoteTrack):
+ (WebCore::LibWebRTCMediaEndpoint::OnAddTrack):
+ (WebCore::LibWebRTCMediaEndpoint::stop):
+ (WebCore::fillEncodingParameters):
+ (WebCore::fillHeaderExtensionParameters):
+ (WebCore::fillCodecParameters):
+ (WebCore::fillRtpParameters):
+ (WebCore::RTCRtpReceiverBackend::getParameters):
+ (WebCore::LibWebRTCMediaEndpoint::getRTCRtpSenderParameters):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::LibWebRTCPeerConnectionBackend::notifyAddedTrack):
+ (WebCore::LibWebRTCPeerConnectionBackend::getParameters):
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/mediastream/mac/RealtimeOutgoingAudioSource.h:
+
2017-03-27 Myles C. Maxfield <mmaxfi...@apple.com>
font variation properties don't need to accept numbers
Modified: trunk/Source/WebCore/DerivedSources.make (214419 => 214420)
--- trunk/Source/WebCore/DerivedSources.make 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/DerivedSources.make 2017-03-27 18:10:45 UTC (rev 214420)
@@ -194,6 +194,7 @@
$(WebCore)/Modules/mediastream/RTCOfferOptions.idl \
$(WebCore)/Modules/mediastream/RTCPeerConnection.idl \
$(WebCore)/Modules/mediastream/RTCPeerConnectionState.idl \
+ $(WebCore)/Modules/mediastream/RTCRtpParameters.idl \
$(WebCore)/Modules/mediastream/RTCRtpReceiver.idl \
$(WebCore)/Modules/mediastream/RTCRtpSender.idl \
$(WebCore)/Modules/mediastream/RTCRtpTransceiver.idl \
Modified: trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/MediaEndpointPeerConnection.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -500,7 +500,7 @@
}
if (!transceiver) {
- auto sender = RTCRtpSender::create(String(mediaDescription.type), Vector<String>(), m_peerConnection.senderClient());
+ auto sender = RTCRtpSender::create(String(mediaDescription.type), Vector<String>(), m_peerConnection.senderBackend());
auto receiver = createReceiver(mediaDescription.mid, mediaDescription.type, mediaDescription.mediaStreamTrackId);
auto newTransceiver = RTCRtpTransceiver::create(WTFMove(sender), WTFMove(receiver));
Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -34,6 +34,7 @@
#if ENABLE(WEB_RTC)
#include "JSDOMPromise.h"
+#include "RTCRtpParameters.h"
#include "RTCSignalingState.h"
namespace WebCore {
@@ -97,6 +98,8 @@
virtual void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
virtual void notifyAddedTrack(RTCRtpSender&) { }
+ virtual RTCRtpParameters getParameters(RTCRtpSender&) const { return { }; }
+
void markAsNeedingNegotiation();
bool isNegotiationNeeded() const { return m_negotiationNeeded; };
void clearNegotiationNeededState() { m_negotiationNeeded = false; };
Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -348,11 +348,14 @@
m_iceConnectionState = RTCIceConnectionState::Closed;
m_signalingState = RTCSignalingState::Closed;
- m_backend->stop();
+ for (RTCRtpReceiver& receiver : m_transceiverSet->receivers())
+ receiver.stop();
for (RTCRtpSender& sender : m_transceiverSet->senders())
sender.stop();
+ m_backend->stop();
+
return true;
}
@@ -522,6 +525,11 @@
m_backend->replaceTrack(sender, withTrack.releaseNonNull(), WTFMove(promise));
}
+RTCRtpParameters RTCPeerConnection::getParameters(RTCRtpSender& sender) const
+{
+ return m_backend->getParameters(sender);
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC)
Modified: trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -58,11 +58,12 @@
struct RTCAnswerOptions;
struct RTCOfferOptions;
+struct RTCRtpParameters;
struct RTCRtpTransceiverInit {
RTCRtpTransceiverDirection direction;
};
-class RTCPeerConnection final : public RefCounted<RTCPeerConnection>, public RTCRtpSenderClient, public EventTargetWithInlineData, public ActiveDOMObject {
+class RTCPeerConnection final : public RefCounted<RTCPeerConnection>, public RTCRtpSender::Backend, public EventTargetWithInlineData, public ActiveDOMObject {
public:
static Ref<RTCPeerConnection> create(ScriptExecutionContext&);
virtual ~RTCPeerConnection();
@@ -134,7 +135,7 @@
void scheduleNegotiationNeededEvent();
- RTCRtpSenderClient& senderClient() { return *this; }
+ RTCRtpSender::Backend& senderBackend() { return *this; }
void fireEvent(Event&);
void disableICECandidateFiltering() { m_backend->disableICECandidateFiltering(); }
@@ -160,8 +161,10 @@
const char* activeDOMObjectName() const final;
bool canSuspendForDocumentSuspension() const final;
- // RTCRtpSenderClient
+ // FIXME: We might want PeerConnectionBackend to be the Backend
+ // RTCRtpSender::Backend
void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final;
+ RTCRtpParameters getParameters(RTCRtpSender&) const final;
void updateConnectionState();
bool doClose();
Added: trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.h (0 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.h (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
+ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
+ * THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#pragma once
+
+#if ENABLE(WEB_RTC)
+
+#include <wtf/Vector.h>
+#include <wtf/text/WTFString.h>
+
+namespace WebCore {
+
+struct RTCRtpParameters {
+
+ struct FecParameters {
+ unsigned long ssrc;
+ };
+
+ struct RtxParameters {
+ unsigned long ssrc;
+ };
+
+ enum class PriorityType { VeryLow, Low, Medium, High };
+ enum class DegradationPreference { MaintainFramerate, MaintainResolution, Balanced };
+ enum class DtxStatus { Disabled, Enabled };
+
+ struct CodecParameters {
+ unsigned short payloadType { 0 };
+ String mimeType;
+ unsigned long clockRate { 0 };
+ unsigned short channels = 1;
+ };
+
+ struct EncodingParameters {
+ unsigned long ssrc { 0 };
+ RtxParameters rtx;
+ FecParameters fec;
+ DtxStatus dtx { DtxStatus::Disabled };
+ bool active { false};
+ PriorityType priority { PriorityType::Medium };
+ unsigned long maxBitrate { 0 };
+ unsigned long maxFramerate { 0 };
+ String rid;
+ double scaleResolutionDownBy { 1 };
+ };
+
+ struct HeaderExtensionParameters {
+ String uri;
+ unsigned short id;
+ };
+
+ String transactionId;
+ Vector<EncodingParameters> encodings;
+ Vector<HeaderExtensionParameters> headerExtensions;
+ Vector<CodecParameters> codecs;
+ DegradationPreference degradationPreference = DegradationPreference::Balanced;
+};
+
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_RTC)
Added: trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.idl (0 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.idl (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpParameters.idl 2017-03-27 18:10:45 UTC (rev 214420)
@@ -0,0 +1,126 @@
+/*
+* Copyright (C) 2017 Apple Inc. All rights reserved.
+*
+* Redistribution and use in source and binary forms, with or without
+* modification, are permitted provided that the following conditions
+* are met:
+* 1. Redistributions of source code must retain the above copyright
+* notice, this list of conditions and the following disclaimer.
+* 2. Redistributions in binary form must reproduce the above copyright
+* notice, this list of conditions and the following disclaimer in the
+* documentation and/or other materials provided with the distribution.
+*
+* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
+* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
+* THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
+* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
+* THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpParameters {
+ DOMString transactionId;
+ sequence<RTCRtpEncodingParameters> encodings;
+ sequence<RTCRtpHeaderExtensionParameters> headerExtensions;
+ // FIXME: Add RTCRtcpParameters rtcp;
+ sequence<RTCRtpCodecParameters> codecs;
+ RTCDegradationPreference degradationPreference = "balanced";
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpCodecParameters {
+ unsigned short payloadType;
+ DOMString mimeType;
+ unsigned long clockRate;
+ unsigned short channels = 1;
+ // FIXME: Add DOMString sdpFmtpLine;
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpEncodingParameters {
+ unsigned long ssrc;
+ RTCRtpRtxParameters rtx;
+ RTCRtpFecParameters fec;
+ RTCDtxStatus dtx;
+ boolean active;
+ RTCPriorityType priority;
+ unsigned long maxBitrate;
+ unsigned long maxFramerate;
+ DOMString rid;
+ double scaleResolutionDownBy = 1;
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpHeaderExtensionParameters {
+ DOMString uri;
+ unsigned short id;
+ // FIXME: Add boolean encrypted;
+};
+
+//[
+// Conditional=WEB_RTC,
+// EnabledAtRuntime=PeerConnection,
+// JSGenerateToJSObject,
+//] dictionary RTCRtcpParameters {
+// DOMString cname;
+// boolean reducedSize;
+//};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpFecParameters {
+ unsigned long ssrc;
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject,
+] dictionary RTCRtpRtxParameters {
+ unsigned long ssrc;
+};
+
+[
+ Conditional=WEB_RTC,
+] enum RTCDegradationPreference {
+ "maintain-framerate",
+ "maintain-resolution",
+ "balanced"
+};
+
+[
+ Conditional=WEB_RTC,
+] enum RTCDtxStatus {
+ "disabled",
+ "enabled"
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+] enum RTCPriorityType {
+ "very-low",
+ "low",
+ "medium",
+ "high"
+};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -33,10 +33,13 @@
#if ENABLE(WEB_RTC)
+#include "RTCRtpParameters.h"
+
namespace WebCore {
-RTCRtpReceiver::RTCRtpReceiver(Ref<MediaStreamTrack>&& track)
+RTCRtpReceiver::RTCRtpReceiver(Ref<MediaStreamTrack>&& track, Backend* backend)
: RTCRtpSenderReceiverBase(WTFMove(track))
+ , m_backend(backend)
{
}
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -32,24 +32,38 @@
#if ENABLE(WEB_RTC)
+#include "RTCRtpParameters.h"
#include "RTCRtpSenderReceiverBase.h"
namespace WebCore {
+
class RTCRtpReceiver : public RTCRtpSenderReceiverBase {
public:
- static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track)
+ class Backend {
+ public:
+ virtual ~Backend() { }
+ virtual RTCRtpParameters getParameters() { return { }; }
+ };
+
+ static Ref<RTCRtpReceiver> create(Ref<MediaStreamTrack>&& track, Backend* backend = nullptr)
{
- return adoptRef(*new RTCRtpReceiver(WTFMove(track)));
+ return adoptRef(*new RTCRtpReceiver(WTFMove(track), backend));
}
+ void stop() { m_backend = nullptr; }
+ // FIXME: We should pass a UniqueRef here.
+ void setBackend(std::unique_ptr<Backend>&& backend) { m_backend = WTFMove(backend); }
+
bool isDispatched() const { return m_isDispatched; }
void setDispatched(bool isDispatched) { m_isDispatched = isDispatched; }
+ RTCRtpParameters getParameters() { return m_backend ? m_backend->getParameters() : RTCRtpParameters(); }
private:
- explicit RTCRtpReceiver(Ref<MediaStreamTrack>&&);
+ explicit RTCRtpReceiver(Ref<MediaStreamTrack>&&, Backend*);
bool m_isDispatched { false };
+ std::unique_ptr<Backend> m_backend;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl 2017-03-27 18:10:45 UTC (rev 214420)
@@ -34,9 +34,9 @@
EnabledAtRuntime=PeerConnection,
] interface RTCRtpReceiver {
readonly attribute MediaStreamTrack track;
+ RTCRtpParameters getParameters();
// FIXME 169662: missing transport
// FIXME 169662: missing rtcpTransport
// FIXME 169662: missing getCapabilities
- // FIXME 169662: missing getParameters
// FIXME 169662: missing getContributingSources
};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -37,23 +37,23 @@
namespace WebCore {
-Ref<RTCRtpSender> RTCRtpSender::create(Ref<MediaStreamTrack>&& track, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
+Ref<RTCRtpSender> RTCRtpSender::create(Ref<MediaStreamTrack>&& track, Vector<String>&& mediaStreamIds, Backend& backend)
{
- auto sender = adoptRef(*new RTCRtpSender(String(track->kind()), WTFMove(mediaStreamIds), client));
+ auto sender = adoptRef(*new RTCRtpSender(String(track->kind()), WTFMove(mediaStreamIds), backend));
sender->setTrack(WTFMove(track));
return sender;
}
-Ref<RTCRtpSender> RTCRtpSender::create(String&& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
+Ref<RTCRtpSender> RTCRtpSender::create(String&& trackKind, Vector<String>&& mediaStreamIds, Backend& backend)
{
- return adoptRef(*new RTCRtpSender(WTFMove(trackKind), WTFMove(mediaStreamIds), client));
+ return adoptRef(*new RTCRtpSender(WTFMove(trackKind), WTFMove(mediaStreamIds), backend));
}
-RTCRtpSender::RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient& client)
+RTCRtpSender::RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds, Backend& backend)
: RTCRtpSenderReceiverBase()
, m_trackKind(WTFMove(trackKind))
, m_mediaStreamIds(WTFMove(mediaStreamIds))
- , m_client(&client)
+ , m_backend(&backend)
{
}
@@ -87,9 +87,16 @@
if (!withTrack && m_track)
m_track->stopProducingData();
- m_client->replaceTrack(*this, WTFMove(withTrack), WTFMove(promise));
+ m_backend->replaceTrack(*this, WTFMove(withTrack), WTFMove(promise));
}
+RTCRtpParameters RTCRtpSender::getParameters()
+{
+ if (isStopped())
+ return { };
+ return m_backend->getParameters(*this);
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC)
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -37,18 +37,18 @@
namespace WebCore {
-class RTCRtpSenderClient {
+class RTCRtpSender : public RTCRtpSenderReceiverBase {
public:
- virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
+ class Backend {
+ public:
+ virtual void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) = 0;
+ virtual RTCRtpParameters getParameters(RTCRtpSender&) const = 0;
+ virtual ~Backend() { }
+ };
- virtual ~RTCRtpSenderClient() { }
-};
+ static Ref<RTCRtpSender> create(Ref<MediaStreamTrack>&&, Vector<String>&& mediaStreamIds, Backend&);
+ static Ref<RTCRtpSender> create(String&& trackKind, Vector<String>&& mediaStreamIds, Backend&);
-class RTCRtpSender : public RTCRtpSenderReceiverBase {
-public:
- static Ref<RTCRtpSender> create(Ref<MediaStreamTrack>&&, Vector<String>&& mediaStreamIds, RTCRtpSenderClient&);
- static Ref<RTCRtpSender> create(String&& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient&);
-
const String& trackId() const { return m_trackId; }
const String& trackKind() const { return m_trackKind; }
@@ -55,20 +55,22 @@
const Vector<String>& mediaStreamIds() const { return m_mediaStreamIds; }
void setMediaStreamIds(Vector<String>&& mediaStreamIds) { m_mediaStreamIds = WTFMove(mediaStreamIds); }
- bool isStopped() const { return !m_client; }
- void stop() { m_client = nullptr; }
+ bool isStopped() const { return !m_backend; }
+ void stop() { m_backend = nullptr; }
void setTrack(Ref<MediaStreamTrack>&&);
void setTrackToNull();
void replaceTrack(RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&);
+ RTCRtpParameters getParameters();
+
private:
- RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds, RTCRtpSenderClient&);
+ RTCRtpSender(String&& trackKind, Vector<String>&& mediaStreamIds, Backend&);
String m_trackId;
String m_trackKind;
Vector<String> m_mediaStreamIds;
- RTCRtpSenderClient* m_client;
+ Backend* m_backend;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl 2017-03-27 18:10:45 UTC (rev 214420)
@@ -38,6 +38,6 @@
// FIXME 169662: missing rtcpTransport
// FIXME 169662: missing getCapabilities
// FIXME 169662: missing setParameters
- // FIXME 169662: missing getParameters
+ RTCRtpParameters getParameters();
Promise<void> replaceTrack(MediaStreamTrack? withTrack);
};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/RTCStatsReport.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -124,7 +124,7 @@
bool ended { false };
bool detached { false };
unsigned long frameWidth { 0 };
- unsigned long frameHeight { 0};
+ unsigned long frameHeight { 0 };
double framesPerSecond { 0 };
unsigned long framesSent { 0 };
unsigned long framesReceived { 0 };
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -160,7 +160,7 @@
m_backend->SetRemoteDescription(&m_setRemoteSessionDescriptionObserver, sessionDescription.release());
}
-void LibWebRTCMediaEndpoint::addTrack(MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
+void LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& track, const Vector<String>& mediaStreamIds)
{
if (!LibWebRTCProvider::factory())
return;
@@ -178,9 +178,8 @@
case RealtimeMediaSource::Type::Audio: {
auto trackSource = RealtimeOutgoingAudioSource::create(source);
auto audioTrack = LibWebRTCProvider::factory()->CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
- trackSource->setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>(audioTrack));
m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
- m_backend->AddTrack(audioTrack.get(), WTFMove(mediaStreams));
+ m_senders.add(&sender, m_backend->AddTrack(audioTrack.get(), WTFMove(mediaStreams)));
return;
}
case RealtimeMediaSource::Type::Video: {
@@ -187,7 +186,7 @@
auto videoSource = RealtimeOutgoingVideoSource::create(source);
auto videoTrack = LibWebRTCProvider::factory()->CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
- m_backend->AddTrack(videoTrack.get(), WTFMove(mediaStreams));
+ m_senders.add(&sender, m_backend->AddTrack(videoTrack.get(), WTFMove(mediaStreams)));
return;
}
case RealtimeMediaSource::Type::None:
@@ -246,6 +245,7 @@
static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
+
if (rtcStats.ssrc.is_defined())
stats.ssrc = *rtcStats.ssrc;
if (rtcStats.associate_stats_id.is_defined())
@@ -276,6 +276,7 @@
static inline void fillInboundRTPStreamStats(RTCStatsReport::InboundRTPStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
{
fillRTCRTPStreamStats(stats, rtcStats);
+
if (rtcStats.packets_received.is_defined())
stats.packetsReceived = *rtcStats.packets_received;
if (rtcStats.bytes_received.is_defined())
@@ -327,6 +328,7 @@
static inline void fillRTCMediaStreamTrackStats(RTCStatsReport::MediaStreamTrackStats& stats, const webrtc::RTCMediaStreamTrackStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
+
if (rtcStats.track_identifier.is_defined())
stats.trackIdentifier = fromStdString(*rtcStats.track_identifier);
if (rtcStats.remote_source.is_defined())
@@ -364,7 +366,7 @@
static inline void fillRTCDataChannelStats(RTCStatsReport::DataChannelStats& stats, const webrtc::RTCDataChannelStats& rtcStats)
{
fillRTCStats(stats, rtcStats);
-
+
if (rtcStats.label.is_defined())
stats.label = fromStdString(*rtcStats.label);
if (rtcStats.protocol.is_defined())
@@ -468,14 +470,25 @@
m_peerConnectionBackend.connection().fireEvent(MediaStreamEvent::create(eventNames().addstreamEvent, false, false, &mediaStream));
}
-void LibWebRTCMediaEndpoint::addRemoteTrack(const webrtc::RtpReceiverInterface& rtcReceiver, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams)
+class RTCRtpReceiverBackend final : public RTCRtpReceiver::Backend {
+public:
+ explicit RTCRtpReceiverBackend(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&& rtcReceiver) : m_rtcReceiver(WTFMove(rtcReceiver)) { }
+private:
+ RTCRtpParameters getParameters() final;
+
+ rtc::scoped_refptr<webrtc::RtpReceiverInterface> m_rtcReceiver;
+};
+
+
+void LibWebRTCMediaEndpoint::addRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&& rtcReceiver, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams)
{
+ ASSERT(rtcReceiver);
RefPtr<RTCRtpReceiver> receiver;
RefPtr<RealtimeMediaSource> remoteSource;
- auto* rtcTrack = rtcReceiver.track().get();
+ auto* rtcTrack = rtcReceiver->track().get();
- switch (rtcReceiver.media_type()) {
+ switch (rtcReceiver->media_type()) {
case cricket::MEDIA_TYPE_DATA:
return;
case cricket::MEDIA_TYPE_AUDIO: {
@@ -496,6 +509,8 @@
}
}
+ receiver->setBackend(std::make_unique<RTCRtpReceiverBackend>(WTFMove(rtcReceiver)));
+
auto* track = receiver->track();
ASSERT(track);
@@ -537,11 +552,10 @@
void LibWebRTCMediaEndpoint::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& streams)
{
- callOnMainThread([protectedThis = makeRef(*this), receiver = WTFMove(receiver), streams] {
+ callOnMainThread([protectedThis = makeRef(*this), receiver = WTFMove(receiver), streams]() mutable {
if (protectedThis->isStopped())
return;
- ASSERT(receiver);
- protectedThis->addRemoteTrack(*receiver, streams);
+ protectedThis->addRemoteTrack(WTFMove(receiver), streams);
});
}
@@ -603,6 +617,7 @@
m_backend->Close();
m_backend = nullptr;
m_streams.clear();
+ m_senders.clear();
}
void LibWebRTCMediaEndpoint::OnRenegotiationNeeded()
@@ -750,6 +765,115 @@
});
}
+static inline RTCRtpParameters::EncodingParameters fillEncodingParameters(const webrtc::RtpEncodingParameters& rtcParameters)
+{
+ RTCRtpParameters::EncodingParameters parameters;
+
+ if (rtcParameters.ssrc)
+ parameters.ssrc = *rtcParameters.ssrc;
+ if (rtcParameters.rtx && rtcParameters.rtx->ssrc)
+ parameters.rtx.ssrc = *rtcParameters.rtx->ssrc;
+ if (rtcParameters.fec && rtcParameters.fec->ssrc)
+ parameters.fec.ssrc = *rtcParameters.fec->ssrc;
+ if (rtcParameters.dtx) {
+ switch (*rtcParameters.dtx) {
+ case webrtc::DtxStatus::DISABLED:
+ parameters.dtx = RTCRtpParameters::DtxStatus::Disabled;
+ break;
+ case webrtc::DtxStatus::ENABLED:
+ parameters.dtx = RTCRtpParameters::DtxStatus::Enabled;
+ }
+ }
+ parameters.active = rtcParameters.active;
+ if (rtcParameters.priority) {
+ switch (*rtcParameters.priority) {
+ case webrtc::PriorityType::VERY_LOW:
+ parameters.priority = RTCRtpParameters::PriorityType::VeryLow;
+ break;
+ case webrtc::PriorityType::LOW:
+ parameters.priority = RTCRtpParameters::PriorityType::Low;
+ break;
+ case webrtc::PriorityType::MEDIUM:
+ parameters.priority = RTCRtpParameters::PriorityType::Medium;
+ break;
+ case webrtc::PriorityType::HIGH:
+ parameters.priority = RTCRtpParameters::PriorityType::High;
+ break;
+ }
+ }
+ if (rtcParameters.max_bitrate_bps)
+ parameters.maxBitrate = *rtcParameters.max_bitrate_bps;
+ if (rtcParameters.max_framerate)
+ parameters.maxFramerate = *rtcParameters.max_framerate;
+ parameters.rid = fromStdString(rtcParameters.rid);
+ parameters.scaleResolutionDownBy = rtcParameters.scale_resolution_down_by;
+
+ return parameters;
+}
+
+static inline RTCRtpParameters::HeaderExtensionParameters fillHeaderExtensionParameters(const webrtc::RtpHeaderExtensionParameters& rtcParameters)
+{
+ RTCRtpParameters::HeaderExtensionParameters parameters;
+
+ parameters.uri = fromStdString(rtcParameters.uri);
+ parameters.id = rtcParameters.id;
+
+ return parameters;
+}
+
+static inline RTCRtpParameters::CodecParameters fillCodecParameters(const webrtc::RtpCodecParameters& rtcParameters)
+{
+ RTCRtpParameters::CodecParameters parameters;
+
+ parameters.payloadType = rtcParameters.payload_type;
+ parameters.mimeType = fromStdString(rtcParameters.mime_type());
+ if (rtcParameters.clock_rate)
+ parameters.clockRate = *rtcParameters.clock_rate;
+ if (rtcParameters.num_channels)
+ parameters.channels = *rtcParameters.num_channels;
+
+ return parameters;
+}
+
+static RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters rtcParameters)
+{
+ RTCRtpParameters parameters;
+
+ parameters.transactionId = fromStdString(rtcParameters.transaction_id);
+ for (auto& rtcEncoding : rtcParameters.encodings)
+ parameters.encodings.append(fillEncodingParameters(rtcEncoding));
+ for (auto& extension : rtcParameters.header_extensions)
+ parameters.headerExtensions.append(fillHeaderExtensionParameters(extension));
+ for (auto& codec : rtcParameters.codecs)
+ parameters.codecs.append(fillCodecParameters(codec));
+
+ switch (rtcParameters.degradation_preference) {
+ case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
+ parameters.degradationPreference = RTCRtpParameters::DegradationPreference::MaintainFramerate;
+ break;
+ case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:
+ parameters.degradationPreference = RTCRtpParameters::DegradationPreference::MaintainResolution;
+ break;
+ case webrtc::DegradationPreference::BALANCED:
+ parameters.degradationPreference = RTCRtpParameters::DegradationPreference::Balanced;
+ break;
+ };
+ return parameters;
+}
+
+RTCRtpParameters RTCRtpReceiverBackend::getParameters()
+{
+ return fillRtpParameters(m_rtcReceiver->GetParameters());
+}
+
+RTCRtpParameters LibWebRTCMediaEndpoint::getRTCRtpSenderParameters(RTCRtpSender& sender)
+{
+ auto rtcSender = m_senders.get(&sender);
+ if (!rtcSender)
+ return { };
+ return fillRtpParameters(rtcSender->GetParameters());
+}
+
} // namespace WebCore
#endif // USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -28,6 +28,7 @@
#include "LibWebRTCProvider.h"
#include "PeerConnectionBackend.h"
+#include "RTCRtpReceiver.h"
#include "RealtimeOutgoingAudioSource.h"
#include "RealtimeOutgoingVideoSource.h"
@@ -79,7 +80,8 @@
RefPtr<RTCSessionDescription> pendingLocalDescription() const;
RefPtr<RTCSessionDescription> pendingRemoteDescription() const;
- void addTrack(MediaStreamTrack&, const Vector<String>&);
+ void addTrack(RTCRtpSender&, MediaStreamTrack&, const Vector<String>&);
+ RTCRtpParameters getRTCRtpSenderParameters(RTCRtpSender&);
private:
LibWebRTCMediaEndpoint(LibWebRTCPeerConnectionBackend&, LibWebRTCProvider&);
@@ -103,7 +105,7 @@
void setRemoteSessionDescriptionSucceeded();
void setRemoteSessionDescriptionFailed(const std::string&);
void addRemoteStream(webrtc::MediaStreamInterface&);
- void addRemoteTrack(const webrtc::RtpReceiverInterface&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&);
+ void addRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&);
void removeRemoteStream(webrtc::MediaStreamInterface&);
void addDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>&&);
@@ -175,6 +177,7 @@
SetLocalSessionDescriptionObserver m_setLocalSessionDescriptionObserver;
SetRemoteSessionDescriptionObserver m_setRemoteSessionDescriptionObserver;
HashMap<webrtc::MediaStreamInterface*, MediaStream*> m_streams;
+ HashMap<RTCRtpSender*, rtc::scoped_refptr<webrtc::RtpSenderInterface>> m_senders;
bool m_isInitiator { false };
};
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2017-03-27 18:10:45 UTC (rev 214420)
@@ -307,7 +307,7 @@
void LibWebRTCPeerConnectionBackend::notifyAddedTrack(RTCRtpSender& sender)
{
ASSERT(sender.track());
- m_endpoint->addTrack(*sender.track(), sender.mediaStreamIds());
+ m_endpoint->addTrack(sender, *sender.track(), sender.mediaStreamIds());
}
void LibWebRTCPeerConnectionBackend::removeRemoteStream(MediaStream* mediaStream)
@@ -364,6 +364,11 @@
}
}
+RTCRtpParameters LibWebRTCPeerConnectionBackend::getParameters(RTCRtpSender& sender) const
+{
+ return m_endpoint->getRTCRtpSenderParameters(sender);
+}
+
} // namespace WebCore
#endif // USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (214419 => 214420)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -70,6 +70,7 @@
RefPtr<RTCSessionDescription> pendingRemoteDescription() const final;
void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromise<void>&&) final;
+ RTCRtpParameters getParameters(RTCRtpSender&) const final;
void emulatePlatformEvent(const String&) final { }
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (214419 => 214420)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2017-03-27 18:10:45 UTC (rev 214420)
@@ -2710,6 +2710,8 @@
5E2C43711BCF0D750001E2BD /* JSRTCRtpTransceiver.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5E2C436D1BCF0D690001E2BD /* JSRTCRtpTransceiver.cpp */; };
5E2C43721BCF0D750001E2BC /* JSRTCRtpReceiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 5E2C436E1BCF0D690001E2BC /* JSRTCRtpReceiver.h */; };
5E2C43721BCF0D750001E2BD /* JSRTCRtpTransceiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 5E2C436E1BCF0D690001E2BD /* JSRTCRtpTransceiver.h */; };
+ 5E2C43731BCF0D750001E2BE /* JSRTCRtpParameters.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5E2C436F1BCF0D690001E2BE /* JSRTCRtpParameters.cpp */; };
+ 5E2C43741BCF0D750001E2BE /* JSRTCRtpParameters.h in Headers */ = {isa = PBXBuildFile; fileRef = 5E2C43701BCF0D690001E2BE /* JSRTCRtpParameters.h */; };
5E2C43731BCF0D750001E2BC /* JSRTCRtpSender.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 5E2C436F1BCF0D690001E2BC /* JSRTCRtpSender.cpp */; };
5E2C43741BCF0D750001E2BC /* JSRTCRtpSender.h in Headers */ = {isa = PBXBuildFile; fileRef = 5E2C43701BCF0D690001E2BC /* JSRTCRtpSender.h */; };
5E2C437B1BCF9A570001E2BC /* RTCPeerConnectionBuiltins.h in Headers */ = {isa = PBXBuildFile; fileRef = 5E2C43761BCF9A0B0001E2BC /* RTCPeerConnectionBuiltins.h */; settings = {ATTRIBUTES = (Private, ); }; };
@@ -9332,6 +9334,8 @@
41A3D58D101C152D00316D07 /* DedicatedWorkerThread.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = DedicatedWorkerThread.h; sourceTree = "<group>"; };
41A48A9C1E83129100D2AC2D /* RTCPeerConnectionState.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = RTCPeerConnectionState.idl; sourceTree = "<group>"; };
41A48A9D1E8312EB00D2AC2D /* RTCPeerConnectionState.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCPeerConnectionState.h; sourceTree = "<group>"; };
+ 41A48AA71E84AEEC00D2AC2D /* RTCRtpParameters.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = RTCRtpParameters.idl; sourceTree = "<group>"; };
+ 41A48AA81E84AF1D00D2AC2D /* RTCRtpParameters.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCRtpParameters.h; sourceTree = "<group>"; };
41ABE6791D0580D5006D862D /* CrossOriginPreflightChecker.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = CrossOriginPreflightChecker.cpp; sourceTree = "<group>"; };
41ABE67A1D0580D5006D862D /* CrossOriginPreflightChecker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CrossOriginPreflightChecker.h; sourceTree = "<group>"; };
41AD75391CEF6BCE00A31486 /* FetchOptions.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = FetchOptions.h; sourceTree = "<group>"; };
@@ -10422,6 +10426,8 @@
5E2C436E1BCF0D690001E2BD /* JSRTCRtpTransceiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSRTCRtpTransceiver.h; sourceTree = "<group>"; };
5E2C436F1BCF0D690001E2BC /* JSRTCRtpSender.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSRTCRtpSender.cpp; sourceTree = "<group>"; };
5E2C43701BCF0D690001E2BC /* JSRTCRtpSender.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSRTCRtpSender.h; sourceTree = "<group>"; };
+ 5E2C436F1BCF0D690001E2BE /* JSRTCRtpParameters.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSRTCRtpParameters.cpp; sourceTree = "<group>"; };
+ 5E2C43701BCF0D690001E2BE /* JSRTCRtpParameters.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSRTCRtpParameters.h; sourceTree = "<group>"; };
5E2C43751BCF9A0B0001E2BC /* RTCPeerConnectionBuiltins.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RTCPeerConnectionBuiltins.cpp; sourceTree = "<group>"; };
5E2C43761BCF9A0B0001E2BC /* RTCPeerConnectionBuiltins.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCPeerConnectionBuiltins.h; sourceTree = "<group>"; };
5E2C43781BCF9A0B0001E2BC /* RTCPeerConnectionInternalsBuiltins.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RTCPeerConnectionInternalsBuiltins.cpp; sourceTree = "<group>"; };
@@ -15958,6 +15964,8 @@
07221B7917CEC32700848E51 /* RTCPeerConnection.idl */,
5E2C43541BCEE30D0001E2BC /* RTCPeerConnection.js */,
5E2C43551BCEE30D0001E2BC /* RTCPeerConnectionInternals.js */,
+ 41A48AA71E84AEEC00D2AC2D /* RTCRtpParameters.idl */,
+ 41A48AA81E84AF1D00D2AC2D /* RTCRtpParameters.h */,
5E2C43561BCEE30D0001E2BC /* RTCRtpReceiver.cpp */,
5E2C43571BCEE30D0001E2BC /* RTCRtpReceiver.h */,
5E2C43581BCEE30D0001E2BC /* RTCRtpReceiver.idl */,
@@ -16196,6 +16204,8 @@
07969DA017D14151007FF842 /* JSRTCPeerConnection.h */,
5E2C436D1BCF0D690001E2BC /* JSRTCRtpReceiver.cpp */,
5E2C436E1BCF0D690001E2BC /* JSRTCRtpReceiver.h */,
+ 5E2C436F1BCF0D690001E2BE /* JSRTCRtpParameters.cpp */,
+ 5E2C43701BCF0D690001E2BE /* JSRTCRtpParameters.h */,
5E2C436F1BCF0D690001E2BC /* JSRTCRtpSender.cpp */,
5E2C43701BCF0D690001E2BC /* JSRTCRtpSender.h */,
5E2C436D1BCF0D690001E2BD /* JSRTCRtpTransceiver.cpp */,
@@ -27856,6 +27866,7 @@
07969DBA17D14151007FF842 /* JSRTCPeerConnection.h in Headers */,
5E2C43721BCF0D750001E2BC /* JSRTCRtpReceiver.h in Headers */,
5E2C43741BCF0D750001E2BC /* JSRTCRtpSender.h in Headers */,
+ 5E2C43741BCF0D750001E2BE /* JSRTCRtpParameters.h in Headers */,
5E2C43721BCF0D750001E2BD /* JSRTCRtpTransceiver.h in Headers */,
07969DBC17D14151007FF842 /* JSRTCSessionDescription.h in Headers */,
07969DC217D14151007FF842 /* JSRTCStatsReport.h in Headers */,
@@ -31822,6 +31833,7 @@
07969DB917D14151007FF842 /* JSRTCPeerConnection.cpp in Sources */,
5E2C43711BCF0D750001E2BC /* JSRTCRtpReceiver.cpp in Sources */,
5E2C43731BCF0D750001E2BC /* JSRTCRtpSender.cpp in Sources */,
+ 5E2C43731BCF0D750001E2BE /* JSRTCRtpParameters.cpp in Sources */,
5E2C43711BCF0D750001E2BD /* JSRTCRtpTransceiver.cpp in Sources */,
07969DBB17D14151007FF842 /* JSRTCSessionDescription.cpp in Sources */,
07969DC117D14151007FF842 /* JSRTCStatsReport.cpp in Sources */,
Modified: trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h (214419 => 214420)
--- trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-03-27 17:50:33 UTC (rev 214419)
+++ trunk/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingAudioSource.h 2017-03-27 18:10:45 UTC (rev 214420)
@@ -48,7 +48,6 @@
static Ref<RealtimeOutgoingAudioSource> create(Ref<RealtimeMediaSource>&& audioSource) { return adoptRef(*new RealtimeOutgoingAudioSource(WTFMove(audioSource))); }
~RealtimeOutgoingAudioSource() { stop(); }
- void setTrack(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& track) { m_track = WTFMove(track); }
void stop();
bool setSource(Ref<RealtimeMediaSource>&&);
@@ -76,7 +75,6 @@
Vector<webrtc::AudioTrackSinkInterface*> m_sinks;
Ref<RealtimeMediaSource> m_audioSource;
- rtc::scoped_refptr<webrtc::AudioTrackInterface> m_track;
Ref<AudioSampleDataSource> m_sampleConverter;
CAAudioStreamDescription m_inputStreamDescription;
CAAudioStreamDescription m_outputStreamDescription;
