Diff
Modified: trunk/LayoutTests/imported/w3c/ChangeLog (237910 => 237911)
--- trunk/LayoutTests/imported/w3c/ChangeLog 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/LayoutTests/imported/w3c/ChangeLog 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1,5 +1,18 @@
2018-11-06 Youenn Fablet <[email protected]>
+ Add support for sender/receiver getCapabilities
+ https://bugs.webkit.org/show_bug.cgi?id=191192
+
+ Reviewed by Eric Carlson.
+
+ * web-platform-tests/webrtc/RTCRtpCapabilities-helper.js:
+ (string_appeared_here.validateRtpCapabilities): Change already made upstream.
+ * web-platform-tests/webrtc/RTCRtpReceiver-getCapabilities-expected.txt:
+ * web-platform-tests/webrtc/RTCRtpSender-getCapabilities-expected.txt:
+ * web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt:
+
+2018-11-06 Youenn Fablet <[email protected]>
+
Support onremovetrack for RTCPeerConnection removed tracks
https://bugs.webkit.org/show_bug.cgi?id=191299
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getCapabilities-expected.txt (237910 => 237911)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getCapabilities-expected.txt 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getCapabilities-expected.txt 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1,5 +1,5 @@
-FAIL RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary RTCRtpReceiver.getCapabilities is not a function. (In 'RTCRtpReceiver.getCapabilities('audio')', 'RTCRtpReceiver.getCapabilities' is undefined)
-FAIL RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary RTCRtpReceiver.getCapabilities is not a function. (In 'RTCRtpReceiver.getCapabilities('video')', 'RTCRtpReceiver.getCapabilities' is undefined)
-FAIL RTCRtpSender.getCapabilities('dummy') should return null RTCRtpReceiver.getCapabilities is not a function. (In 'RTCRtpReceiver.getCapabilities('dummy')', 'RTCRtpReceiver.getCapabilities' is undefined)
+PASS RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary
+PASS RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary
+PASS RTCRtpSender.getCapabilities('dummy') should return null
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getCapabilities-expected.txt (237910 => 237911)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getCapabilities-expected.txt 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-getCapabilities-expected.txt 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1,5 +1,5 @@
-FAIL RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
-FAIL RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('video')', 'RTCRtpSender.getCapabilities' is undefined)
-FAIL RTCRtpSender.getCapabilities('dummy') should return null RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('dummy')', 'RTCRtpSender.getCapabilities' is undefined)
+PASS RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary
+PASS RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary
+PASS RTCRtpSender.getCapabilities('dummy') should return null
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt (237910 => 237911)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver-setCodecPreferences-expected.txt 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1,11 +1,11 @@
-FAIL setCodecPreferences() on audio transceiver with codecs returned from RTCRtpSender.getCapabilities('audio') should succeed RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
-FAIL setCodecPreferences() on video transceiver with codecs returned from RTCRtpReceiver.getCapabilities('video') should succeed RTCRtpReceiver.getCapabilities is not a function. (In 'RTCRtpReceiver.getCapabilities('video')', 'RTCRtpReceiver.getCapabilities' is undefined)
-FAIL setCodecPreferences() with both sender receiver codecs combined should succeed RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
+FAIL setCodecPreferences() on audio transceiver with codecs returned from RTCRtpSender.getCapabilities('audio') should succeed transceiver.setCodecPreferences is not a function. (In 'transceiver.setCodecPreferences(capabilities.codecs)', 'transceiver.setCodecPreferences' is undefined)
+FAIL setCodecPreferences() on video transceiver with codecs returned from RTCRtpReceiver.getCapabilities('video') should succeed transceiver.setCodecPreferences is not a function. (In 'transceiver.setCodecPreferences(capabilities.codecs)', 'transceiver.setCodecPreferences' is undefined)
+FAIL setCodecPreferences() with both sender receiver codecs combined should succeed transceiver.setCodecPreferences is not a function. (In 'transceiver.setCodecPreferences([...capabilities1.codecs, ... capabilities2.codecs])', 'transceiver.setCodecPreferences' is undefined)
FAIL setCodecPreferences([]) should succeed transceiver.setCodecPreferences is not a function. (In 'transceiver.setCodecPreferences([])', 'transceiver.setCodecPreferences' is undefined)
-FAIL setCodecPreferences() with reordered codecs should succeed RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
-FAIL setCodecPreferences() on audio transceiver with codecs returned from getCapabilities('video') should throw InvalidAccessError RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('video')', 'RTCRtpSender.getCapabilities' is undefined)
+FAIL setCodecPreferences() with reordered codecs should succeed transceiver.setCodecPreferences is not a function. (In 'transceiver.setCodecPreferences(codecs)', 'transceiver.setCodecPreferences' is undefined)
+FAIL setCodecPreferences() on audio transceiver with codecs returned from getCapabilities('video') should throw InvalidAccessError Test bug: unrecognized DOMException code "() => transceiver.setCodecPreferences(capabilities.codecs)" passed to assert_throws()
FAIL setCodecPreferences() with user defined codec should throw InvalidAccessError Test bug: unrecognized DOMException code "() => transceiver.setCodecPreferences(codecs)" passed to assert_throws()
-FAIL setCodecPreferences() with user defined codec together with codecs returned from getCapabilities() should throw InvalidAccessError RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
-FAIL setCodecPreferences() with modified codecs returned from getCapabilities() should throw InvalidAccessError RTCRtpSender.getCapabilities is not a function. (In 'RTCRtpSender.getCapabilities('audio')', 'RTCRtpSender.getCapabilities' is undefined)
+FAIL setCodecPreferences() with user defined codec together with codecs returned from getCapabilities() should throw InvalidAccessError Test bug: unrecognized DOMException code "() => transceiver.setCodecPreferences(codecs)" passed to assert_throws()
+FAIL setCodecPreferences() with modified codecs returned from getCapabilities() should throw InvalidAccessError Test bug: unrecognized DOMException code "() => transceiver.setCodecPreferences(codecs)" passed to assert_throws()
Modified: trunk/Source/WebCore/CMakeLists.txt (237910 => 237911)
--- trunk/Source/WebCore/CMakeLists.txt 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/CMakeLists.txt 2018-11-07 04:01:49 UTC (rev 237911)
@@ -333,6 +333,7 @@
Modules/mediastream/RTCPeerConnectionIceEvent.idl
Modules/mediastream/RTCPeerConnectionState.idl
Modules/mediastream/RTCPriorityType.idl
+ Modules/mediastream/RTCRtpCapabilities.idl
Modules/mediastream/RTCRtpCodecParameters.idl
Modules/mediastream/RTCRtpContributingSource.idl
Modules/mediastream/RTCRtpEncodingParameters.idl
Modified: trunk/Source/WebCore/ChangeLog (237910 => 237911)
--- trunk/Source/WebCore/ChangeLog 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/ChangeLog 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1,5 +1,47 @@
2018-11-06 Youenn Fablet <[email protected]>
+ Add support for sender/receiver getCapabilities
+ https://bugs.webkit.org/show_bug.cgi?id=191192
+
+ Reviewed by Eric Carlson.
+
+ Expose sender/receiver RTCRtpCapabilities to JS.
+ Add corresponding IDL and plumbing down to libwebrtc peer connection factory.
+ Covered by rebased tests.
+
+ * CMakeLists.txt:
+ * DerivedSources.make:
+ * Modules/mediastream/PeerConnectionBackend.cpp:
+ (WebCore::PeerConnectionBackend::receiverCapabilities):
+ (WebCore::PeerConnectionBackend::senderCapabilities):
+ * Modules/mediastream/PeerConnectionBackend.h:
+ * Modules/mediastream/RTCRtpCapabilities.idl: Added.
+ * Modules/mediastream/RTCRtpReceiver.cpp:
+ (WebCore::RTCRtpReceiver::getCapabilities):
+ * Modules/mediastream/RTCRtpReceiver.h:
+ * Modules/mediastream/RTCRtpReceiver.idl:
+ * Modules/mediastream/RTCRtpSender.cpp:
+ (WebCore::RTCRtpSender::getCapabilities):
+ * Modules/mediastream/RTCRtpSender.h:
+ * Modules/mediastream/RTCRtpSender.idl:
+ * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+ (WebCore::PeerConnectionBackend::receiverCapabilities):
+ (WebCore::PeerConnectionBackend::senderCapabilities):
+ * Sources.txt:
+ * WebCore.xcodeproj/project.pbxproj:
+ * platform/mediastream/RTCPMuxPolicy.h:
+ * platform/mediastream/RTCRtpCapabilities.h: Added.
+ * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
+ (WebCore::typeFromKind):
+ (WebCore::fromStdString):
+ (WebCore::toChannels):
+ (WebCore::toRTCRtpCapabilities):
+ (WebCore::LibWebRTCProvider::receiverCapabilities):
+ (WebCore::LibWebRTCProvider::senderCapabilities):
+ * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
+
+2018-11-06 Youenn Fablet <[email protected]>
+
Calling sender.replaceTrack() twice produces a new transceiver and its corresponding m= section
https://bugs.webkit.org/show_bug.cgi?id=191261
Modified: trunk/Source/WebCore/DerivedSources.make (237910 => 237911)
--- trunk/Source/WebCore/DerivedSources.make 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/DerivedSources.make 2018-11-07 04:01:49 UTC (rev 237911)
@@ -254,6 +254,7 @@
$(WebCore)/Modules/mediastream/RTCOfferOptions.idl \
$(WebCore)/Modules/mediastream/RTCPeerConnection.idl \
$(WebCore)/Modules/mediastream/RTCPeerConnectionState.idl \
+ $(WebCore)/Modules/mediastream/RTCRtpCapabilities.idl \
$(WebCore)/Modules/mediastream/RTCRtpCodecParameters.idl \
$(WebCore)/Modules/mediastream/RTCRtpContributingSource.idl \
$(WebCore)/Modules/mediastream/RTCRtpEncodingParameters.idl \
Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.cpp 2018-11-07 04:01:49 UTC (rev 237911)
@@ -42,6 +42,7 @@
#include "RTCIceCandidate.h"
#include "RTCPeerConnection.h"
#include "RTCPeerConnectionIceEvent.h"
+#include "RTCRtpCapabilities.h"
#include "RuntimeEnabledFeatures.h"
#include <wtf/text/StringBuilder.h>
#include <wtf/text/StringConcatenateNumbers.h>
@@ -57,6 +58,18 @@
}
CreatePeerConnectionBackend PeerConnectionBackend::create = createNoPeerConnectionBackend;
+
+std::optional<RTCRtpCapabilities> PeerConnectionBackend::receiverCapabilities(ScriptExecutionContext&, const String&)
+{
+ ASSERT_NOT_REACHED();
+ return { };
+}
+
+std::optional<RTCRtpCapabilities> PeerConnectionBackend::senderCapabilities(ScriptExecutionContext&, const String&)
+{
+ ASSERT_NOT_REACHED();
+ return { };
+}
#endif
PeerConnectionBackend::PeerConnectionBackend(RTCPeerConnection& peerConnection)
Modified: trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/PeerConnectionBackend.h 2018-11-07 04:01:49 UTC (rev 237911)
@@ -79,6 +79,9 @@
public:
WEBCORE_EXPORT static CreatePeerConnectionBackend create;
+ static std::optional<RTCRtpCapabilities> receiverCapabilities(ScriptExecutionContext&, const String& kind);
+ static std::optional<RTCRtpCapabilities> senderCapabilities(ScriptExecutionContext&, const String& kind);
+
explicit PeerConnectionBackend(RTCPeerConnection&);
virtual ~PeerConnectionBackend() = default;
Added: trunk/Source/WebCore/Modules/mediastream/RTCRtpCapabilities.idl (0 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpCapabilities.idl (rev 0)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpCapabilities.idl 2018-11-07 04:01:49 UTC (rev 237911)
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2018 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 RTCRtpCodecCapability {
+ required DOMString mimeType;
+ required unsigned long clockRate;
+ unsigned short channels;
+ DOMString sdpFmtpLine;
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject
+] dictionary RTCRtpHeaderExtensionCapability {
+ DOMString uri;
+};
+
+[
+ Conditional=WEB_RTC,
+ EnabledAtRuntime=PeerConnection,
+ JSGenerateToJSObject
+] dictionary RTCRtpCapabilities {
+ required sequence<RTCRtpCodecCapability> codecs;
+ required sequence<RTCRtpHeaderExtensionCapability> headerExtensions;
+};
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.cpp 2018-11-07 04:01:49 UTC (rev 237911)
@@ -32,7 +32,9 @@
#include "RTCRtpReceiver.h"
#if ENABLE(WEB_RTC)
+
#include "PeerConnectionBackend.h"
+#include "RTCRtpCapabilities.h"
namespace WebCore {
@@ -61,6 +63,11 @@
m_connection->getStats(*this, WTFMove(promise));
}
+std::optional<RTCRtpCapabilities> RTCRtpReceiver::getCapabilities(ScriptExecutionContext& context, const String& kind)
+{
+ return PeerConnectionBackend::receiverCapabilities(context, kind);
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC)
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.h 2018-11-07 04:01:49 UTC (rev 237911)
@@ -40,6 +40,7 @@
namespace WebCore {
class PeerConnectionBackend;
+struct RTCRtpCapabilities;
class RTCRtpReceiver : public RefCounted<RTCRtpReceiver>, public ScriptWrappable {
public:
@@ -48,6 +49,8 @@
return adoptRef(*new RTCRtpReceiver(connection, WTFMove(track), WTFMove(backend)));
}
+ static std::optional<RTCRtpCapabilities> getCapabilities(ScriptExecutionContext&, const String& kind);
+
void stop();
void setBackend(std::unique_ptr<RTCRtpReceiverBackend>&& backend) { m_backend = WTFMove(backend); }
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpReceiver.idl 2018-11-07 04:01:49 UTC (rev 237911)
@@ -35,10 +35,10 @@
ImplementationLacksVTable
] interface RTCRtpReceiver {
readonly attribute MediaStreamTrack track;
- RTCRtpParameters getParameters();
// FIXME 169662: missing transport
// FIXME 169662: missing rtcpTransport
- // FIXME 169662: missing getCapabilities
+ [CallWith=ScriptExecutionContext] static RTCRtpCapabilities? getCapabilities(DOMString kind);
+ RTCRtpParameters getParameters();
sequence<RTCRtpContributingSource> getContributingSources();
sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
Promise<RTCStatsReport> getStats();
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.cpp 2018-11-07 04:01:49 UTC (rev 237911)
@@ -33,6 +33,7 @@
#if ENABLE(WEB_RTC)
+#include "RTCRtpCapabilities.h"
#include "RuntimeEnabledFeatures.h"
namespace WebCore {
@@ -121,6 +122,11 @@
m_connection->getStats(*this, WTFMove(promise));
}
+std::optional<RTCRtpCapabilities> RTCRtpSender::getCapabilities(ScriptExecutionContext& context, const String& kind)
+{
+ return PeerConnectionBackend::senderCapabilities(context, kind);
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_RTC)
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.h 2018-11-07 04:01:49 UTC (rev 237911)
@@ -40,6 +40,7 @@
namespace WebCore {
class PeerConnectionBackend;
+struct RTCRtpCapabilities;
class RTCRtpSender : public RefCounted<RTCRtpSender>, public ScriptWrappable {
public:
@@ -46,6 +47,8 @@
static Ref<RTCRtpSender> create(PeerConnectionBackend&, Ref<MediaStreamTrack>&&, Vector<String>&& mediaStreamIds, std::unique_ptr<RTCRtpSenderBackend>&&);
static Ref<RTCRtpSender> create(PeerConnectionBackend&, String&& trackKind, Vector<String>&& mediaStreamIds, std::unique_ptr<RTCRtpSenderBackend>&&);
+ static std::optional<RTCRtpCapabilities> getCapabilities(ScriptExecutionContext&, const String& kind);
+
MediaStreamTrack* track() { return m_track.get(); }
const String& trackId() const { return m_trackId; }
Modified: trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/RTCRtpSender.idl 2018-11-07 04:01:49 UTC (rev 237911)
@@ -37,8 +37,8 @@
readonly attribute MediaStreamTrack? track;
// FIXME 169662: missing transport
// FIXME 169662: missing rtcpTransport
- // FIXME 169662: missing getCapabilities
// FIXME 169662: missing setStreams
+ [CallWith=ScriptExecutionContext] static RTCRtpCapabilities? getCapabilities(DOMString kind);
RTCRtpSendParameters getParameters();
Promise<void> setParameters(RTCRtpSendParameters parameters);
[CallWith=ScriptExecutionContext] Promise<void> replaceTrack(MediaStreamTrack? withTrack);
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (237910 => 237911)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp 2018-11-07 04:01:49 UTC (rev 237911)
@@ -38,6 +38,7 @@
#include "Page.h"
#include "RTCIceCandidate.h"
#include "RTCPeerConnection.h"
+#include "RTCRtpCapabilities.h"
#include "RTCRtpReceiver.h"
#include "RTCSessionDescription.h"
#include "RealtimeIncomingAudioSource.h"
@@ -65,6 +66,22 @@
CreatePeerConnectionBackend PeerConnectionBackend::create = createLibWebRTCPeerConnectionBackend;
+std::optional<RTCRtpCapabilities> PeerConnectionBackend::receiverCapabilities(ScriptExecutionContext& context, const String& kind)
+{
+ auto* page = downcast<Document>(context).page();
+ if (!page)
+ return { };
+ return page->libWebRTCProvider().receiverCapabilities(kind);
+}
+
+std::optional<RTCRtpCapabilities> PeerConnectionBackend::senderCapabilities(ScriptExecutionContext& context, const String& kind)
+{
+ auto* page = downcast<Document>(context).page();
+ if (!page)
+ return { };
+ return page->libWebRTCProvider().senderCapabilities(kind);
+}
+
LibWebRTCPeerConnectionBackend::LibWebRTCPeerConnectionBackend(RTCPeerConnection& peerConnection, LibWebRTCProvider& provider)
: PeerConnectionBackend(peerConnection)
, m_endpoint(LibWebRTCMediaEndpoint::create(*this, provider))
Modified: trunk/Source/WebCore/Sources.txt (237910 => 237911)
--- trunk/Source/WebCore/Sources.txt 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/Sources.txt 2018-11-07 04:01:49 UTC (rev 237911)
@@ -2936,6 +2936,7 @@
JSRTCPeerConnectionIceEvent.cpp
JSRTCPeerConnectionState.cpp
JSRTCPriorityType.cpp
+JSRTCRtpCapabilities.cpp
JSRTCRtpCodecParameters.cpp
JSRTCRtpContributingSource.cpp
JSRTCRtpEncodingParameters.cpp
Modified: trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj (237910 => 237911)
--- trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/WebCore.xcodeproj/project.pbxproj 2018-11-07 04:01:49 UTC (rev 237911)
@@ -1056,6 +1056,7 @@
4129C9AB1F59C573009D7403 /* ReadableStreamSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 418C395B1C8F0A610051C8A3 /* ReadableStreamSource.h */; settings = {ATTRIBUTES = (Private, ); }; };
4129C9AF1F59CF5B009D7403 /* ReadableStreamSink.h in Headers */ = {isa = PBXBuildFile; fileRef = 4129C9801F5861C7009D7403 /* ReadableStreamSink.h */; settings = {ATTRIBUTES = (Private, ); }; };
4129DF861BB5B80C00322A16 /* JSReadableStreamPrivateConstructors.h in Headers */ = {isa = PBXBuildFile; fileRef = 4129DF841BB5B7F700322A16 /* JSReadableStreamPrivateConstructors.h */; settings = {ATTRIBUTES = (Private, ); }; };
+ 412DE4B8219285C00075F3A7 /* RTCRtpCapabilities.h in Headers */ = {isa = PBXBuildFile; fileRef = 412DE4B6219285BF0075F3A7 /* RTCRtpCapabilities.h */; };
4133CB8B20F80E9900E89B11 /* MediaStreamAudioSourceCocoa.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 4133CB8920F80E8600E89B11 /* MediaStreamAudioSourceCocoa.cpp */; };
41380C271F3436AC00155FDA /* DOMCache.h in Headers */ = {isa = PBXBuildFile; fileRef = 41380C251F34369A00155FDA /* DOMCache.h */; };
41380C291F3436AC00155FDA /* DOMCacheStorage.h in Headers */ = {isa = PBXBuildFile; fileRef = 41380C221F34369000155FDA /* DOMCacheStorage.h */; };
@@ -7265,6 +7266,7 @@
4129DF831BB5B7F700322A16 /* JSReadableStreamPrivateConstructors.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSReadableStreamPrivateConstructors.cpp; sourceTree = "<group>"; };
4129DF841BB5B7F700322A16 /* JSReadableStreamPrivateConstructors.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSReadableStreamPrivateConstructors.h; sourceTree = "<group>"; };
412BA5FE218C651800365474 /* RTCPMuxPolicy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCPMuxPolicy.h; sourceTree = "<group>"; };
+ 412DE4B6219285BF0075F3A7 /* RTCRtpCapabilities.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCRtpCapabilities.h; sourceTree = "<group>"; };
413015D51C7B570400091C6E /* FetchResponse.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = FetchResponse.cpp; sourceTree = "<group>"; };
413015D51C7B570400091C6F /* FetchBodySource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = FetchBodySource.cpp; sourceTree = "<group>"; };
413015D61C7B570400091C6E /* FetchResponse.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = FetchResponse.h; sourceTree = "<group>"; };
@@ -7274,6 +7276,7 @@
4131F3B41F955BC30059995A /* ExtendableEventInit.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ExtendableEventInit.h; sourceTree = "<group>"; };
4131F3B51F955BC50059995A /* ExtendableEventInit.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = ExtendableEventInit.idl; sourceTree = "<group>"; };
4133CB8920F80E8600E89B11 /* MediaStreamAudioSourceCocoa.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = MediaStreamAudioSourceCocoa.cpp; sourceTree = "<group>"; };
+ 41369E55218C76E300792E29 /* RTCRtpCapabilities.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = RTCRtpCapabilities.idl; sourceTree = "<group>"; };
41380C201F34368A00155FDA /* DOMCache.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = DOMCache.cpp; path = Modules/cache/DOMCache.cpp; sourceTree = SOURCE_ROOT; };
41380C211F34368D00155FDA /* DOMCacheStorage.cpp */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = DOMCacheStorage.cpp; path = Modules/cache/DOMCacheStorage.cpp; sourceTree = SOURCE_ROOT; };
41380C221F34369000155FDA /* DOMCacheStorage.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = DOMCacheStorage.h; path = Modules/cache/DOMCacheStorage.h; sourceTree = SOURCE_ROOT; };
@@ -15458,6 +15461,7 @@
41A48A9C1E83129100D2AC2D /* RTCPeerConnectionState.idl */,
41FCB75C214866FE0038ADC6 /* RTCPriorityType.h */,
41E593FD214865A900D3CB61 /* RTCPriorityType.idl */,
+ 41369E55218C76E300792E29 /* RTCRtpCapabilities.idl */,
41FCB75F214866FF0038ADC6 /* RTCRtpCodecParameters.h */,
41FCB759214865D30038ADC6 /* RTCRtpCodecParameters.idl */,
414AD3FF21498D3000521676 /* RTCRtpCodingParameters.h */,
@@ -15570,6 +15574,7 @@
07221BAA17CF0AD400848E51 /* RTCPeerConnectionHandlerClient.h */,
41A48A9D1E8312EB00D2AC2D /* RTCPeerConnectionState.h */,
412BA5FE218C651800365474 /* RTCPMuxPolicy.h */,
+ 412DE4B6219285BF0075F3A7 /* RTCRtpCapabilities.h */,
31EB54DD1E7DC74400C1623B /* RTCRtpTransceiverDirection.h */,
313591041E7DDC6000F30630 /* RTCSdpType.h */,
07221BAB17CF0AD400848E51 /* RTCSessionDescriptionDescriptor.cpp */,
@@ -28590,6 +28595,7 @@
51E399021D6E4750009C8831 /* GameControllerGamepadProvider.h in Headers */,
516C62211950D48700337E75 /* GamepadEvent.h in Headers */,
51A9D9EA195B931F001B2B5C /* GamepadManager.h in Headers */,
+ 412DE4B8219285C00075F3A7 /* RTCRtpCapabilities.h in Headers */,
515BE1921D54F5FB00DD7C68 /* GamepadProvider.h in Headers */,
515BE1931D54F5FB00DD7C68 /* GamepadProviderClient.h in Headers */,
8EC6C963201A251600FBFA53 /* GapLength.h in Headers */,
Added: trunk/Source/WebCore/platform/mediastream/RTCRtpCapabilities.h (0 => 237911)
--- trunk/Source/WebCore/platform/mediastream/RTCRtpCapabilities.h (rev 0)
+++ trunk/Source/WebCore/platform/mediastream/RTCRtpCapabilities.h 2018-11-07 04:01:49 UTC (rev 237911)
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2018 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 RTCRtpCapabilities {
+ struct CodecCapability {
+ String mimeType;
+ uint32_t clockRate { 0 };
+ std::optional<uint16_t> channels;
+ String sdpFmtpLine;
+ };
+ struct HeaderExtensionCapability {
+ String uri;
+ };
+
+ Vector<CodecCapability> codecs;
+ Vector<HeaderExtensionCapability> headerExtensions;
+};
+
+}; // namespace WebCore
+
+#endif
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp (237910 => 237911)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp 2018-11-07 04:01:49 UTC (rev 237911)
@@ -29,6 +29,7 @@
#if USE(LIBWEBRTC)
#include "LibWebRTCAudioModule.h"
#include "Logging.h"
+#include "RTCRtpCapabilities.h"
#include <dlfcn.h>
ALLOW_UNUSED_PARAMETERS_BEGIN
@@ -280,6 +281,68 @@
return *factoryAndThreads.certificateGenerator;
}
+static inline std::optional<cricket::MediaType> typeFromKind(const String& kind)
+{
+ if (kind == "audio"_s)
+ return cricket::MediaType::MEDIA_TYPE_AUDIO;
+ if (kind == "video"_s)
+ return cricket::MediaType::MEDIA_TYPE_VIDEO;
+ return { };
+}
+
+static inline String fromStdString(const std::string& value)
+{
+ return String::fromUTF8(value.data(), value.length());
+}
+
+static inline std::optional<uint16_t> toChannels(absl::optional<int> numChannels)
+{
+ if (!numChannels)
+ return { };
+ return static_cast<uint32_t>(*numChannels);
+}
+
+static inline RTCRtpCapabilities toRTCRtpCapabilities(const webrtc::RtpCapabilities& rtpCapabilities)
+{
+ RTCRtpCapabilities capabilities;
+
+ capabilities.codecs.reserveInitialCapacity(rtpCapabilities.codecs.size());
+ for (auto& codec : rtpCapabilities.codecs)
+ capabilities.codecs.uncheckedAppend(RTCRtpCapabilities::CodecCapability { fromStdString(codec.mime_type()), static_cast<uint32_t>(codec.clock_rate ? *codec.clock_rate : 0), toChannels(codec.num_channels), { } });
+
+ capabilities.headerExtensions.reserveInitialCapacity(rtpCapabilities.header_extensions.size());
+ for (auto& header : rtpCapabilities.header_extensions)
+ capabilities.headerExtensions.uncheckedAppend(RTCRtpCapabilities::HeaderExtensionCapability { fromStdString(header.uri) });
+
+ return capabilities;
+}
+
+std::optional<RTCRtpCapabilities> LibWebRTCProvider::receiverCapabilities(const String& kind)
+{
+ auto mediaType = typeFromKind(kind);
+ if (!mediaType)
+ return { };
+
+ auto* factory = this->factory();
+ if (!factory)
+ return { };
+
+ return toRTCRtpCapabilities(factory->GetRtpReceiverCapabilities(*mediaType));
+}
+
+std::optional<RTCRtpCapabilities> LibWebRTCProvider::senderCapabilities(const String& kind)
+{
+ auto mediaType = typeFromKind(kind);
+ if (!mediaType)
+ return { };
+
+ auto* factory = this->factory();
+ if (!factory)
+ return { };
+
+ return toRTCRtpCapabilities(factory->GetRtpSenderCapabilities(*mediaType));
+}
+
#endif // USE(LIBWEBRTC)
} // namespace WebCore
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h (237910 => 237911)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h 2018-11-07 03:23:59 UTC (rev 237910)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h 2018-11-07 04:01:49 UTC (rev 237911)
@@ -59,6 +59,7 @@
namespace WebCore {
class LibWebRTCAudioModule;
+struct RTCRtpCapabilities;
enum class MDNSRegisterError { NotImplemented, BadParameter, DNSSD, Internal, Timeout };
@@ -109,6 +110,9 @@
rtc::RTCCertificateGenerator& certificateGenerator();
+ std::optional<RTCRtpCapabilities> receiverCapabilities(const String& kind);
+ std::optional<RTCRtpCapabilities> senderCapabilities(const String& kind);
+
protected:
LibWebRTCProvider() = default;
