[webkit-changes] [235722] trunk/Source/WebCore

Wed, 05 Sep 2018 22:10:29 -0700

Title: [235722] trunk/Source/WebCore
Revision
235722
Author
[email protected]
Date
2018-09-05 22:08:43 -0700 (Wed, 05 Sep 2018)

Log Message

Move ownership of outgoing source to RTCRtpSender backend
https://bugs.webkit.org/show_bug.cgi?id=189310

Reviewed by Alex Christensen.

RTCRtpSender should own the source so that it can replace/stop it.
Since this is libwebrtc specific, the source is actually owned by the backend.
Simplified replaceTrack a bit based on that.

No change of behavior.

* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::addTrack):
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
(WebCore::LibWebRTCPeerConnectionBackend::doStop):
(WebCore::updateTrackSource):
(WebCore::LibWebRTCPeerConnectionBackend::replaceTrack):
(WebCore::LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources):
* Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
* Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:

Modified Paths

Diff

Modified: trunk/Source/WebCore/ChangeLog (235721 => 235722)


--- trunk/Source/WebCore/ChangeLog	2018-09-06 04:07:53 UTC (rev 235721)
+++ trunk/Source/WebCore/ChangeLog	2018-09-06 05:08:43 UTC (rev 235722)
@@ -1,3 +1,26 @@
+2018-09-05  Youenn Fablet  <[email protected]>
+
+        Move ownership of outgoing source to RTCRtpSender backend
+        https://bugs.webkit.org/show_bug.cgi?id=189310
+
+        Reviewed by Alex Christensen.
+
+        RTCRtpSender should own the source so that it can replace/stop it.
+        Since this is libwebrtc specific, the source is actually owned by the backend.
+        Simplified replaceTrack a bit based on that.
+
+        No change of behavior.
+
+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
+        (WebCore::LibWebRTCPeerConnectionBackend::doStop):
+        (WebCore::updateTrackSource):
+        (WebCore::LibWebRTCPeerConnectionBackend::replaceTrack):
+        (WebCore::LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources):
+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
+        * Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h:
+
 2018-09-05  Brent Fulgham  <[email protected]>
 
         The width of an empty or nullptr TextRun should be zero

Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (235721 => 235722)


--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp	2018-09-06 04:07:53 UTC (rev 235721)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp	2018-09-06 05:08:43 UTC (rev 235722)
@@ -214,9 +214,9 @@
 
     switch (track.privateTrack().type()) {
     case RealtimeMediaSource::Type::Audio: {
-        auto trackSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
-        auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
-        m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
+        auto audioSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
+        auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), audioSource.ptr());
+        sender.setSource(WTFMove(audioSource));
         auto rtpSender = m_backend->AddTrack(audioTrack.get(), WTFMove(ids));
         if (!rtpSender.ok())
             return false;
@@ -226,7 +226,7 @@
     case RealtimeMediaSource::Type::Video: {
         auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
         auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
-        m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
+        sender.setSource(WTFMove(videoSource));
         auto rtpSender = m_backend->AddTrack(videoTrack.get(), WTFMove(ids));
         if (!rtpSender.ok())
             return false;

Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp (235721 => 235722)


--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp	2018-09-06 04:07:53 UTC (rev 235721)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp	2018-09-06 05:08:43 UTC (rev 235722)
@@ -174,15 +174,7 @@
 
 void LibWebRTCPeerConnectionBackend::doStop()
 {
-    for (auto& source : m_audioSources)
-        source->stop();
-    for (auto& source : m_videoSources)
-        source->stop();
-
     m_endpoint->stop();
-
-    m_audioSources.clear();
-    m_videoSources.clear();
     m_pendingReceivers.clear();
 }
 
@@ -208,16 +200,6 @@
     addIceCandidateSucceeded();
 }
 
-void LibWebRTCPeerConnectionBackend::addAudioSource(Ref<RealtimeOutgoingAudioSource>&& source)
-{
-    m_audioSources.append(WTFMove(source));
-}
-
-void LibWebRTCPeerConnectionBackend::addVideoSource(Ref<RealtimeOutgoingVideoSource>&& source)
-{
-    m_videoSources.append(WTFMove(source));
-}
-
 static inline Ref<RTCRtpReceiver> createReceiverForSource(ScriptExecutionContext& context, Ref<RealtimeMediaSource>&& source)
 {
     auto remoteTrackPrivate = MediaStreamTrackPrivate::create(WTFMove(source), String { source->id() });
@@ -395,25 +377,12 @@
 static inline bool updateTrackSource(Source& source, MediaStreamTrack* track)
 {
     if (!track) {
-        source->stop();
+        source.stop();
         return true;
     }
-    return source->setSource(track->privateTrack());
+    return source.setSource(track->privateTrack());
 }
 
-template<typename Source>
-static inline bool tryUpdatingTrackSource(MediaStreamTrack& currentTrack, MediaStreamTrack* newTrack, const Vector<Source>& sources)
-{
-    for (auto& source : sources) {
-        if (&source->source() == &currentTrack.privateTrack()) {
-            if (!updateTrackSource(source, newTrack))
-                return false;
-            return true;
-        }
-    }
-    return false;
-}
-
 void LibWebRTCPeerConnectionBackend::replaceTrack(RTCRtpSender& sender, RefPtr<MediaStreamTrack>&& track, DOMPromiseDeferred<void>&& promise)
 {
     auto* currentTrack = sender.track();
@@ -426,13 +395,13 @@
             promise.reject(InvalidModificationError);
             break;
         case RealtimeMediaSource::Type::Audio:
-            if (!tryUpdatingTrackSource(*currentTrack, track.get(), m_audioSources)) {
+            if (!updateTrackSource(*backendFromRTPSender(sender).audioSource(), track.get())) {
                 promise.reject(InvalidModificationError);
                 return;
             }
             break;
         case RealtimeMediaSource::Type::Video:
-            if (!tryUpdatingTrackSource(*currentTrack, track.get(), m_videoSources)) {
+            if (!updateTrackSource(*backendFromRTPSender(sender).videoSource(), track.get())) {
                 promise.reject(InvalidModificationError);
                 return;
             }
@@ -468,8 +437,10 @@
 
 void LibWebRTCPeerConnectionBackend::applyRotationForOutgoingVideoSources()
 {
-    for (auto& source : m_videoSources)
-        source->setApplyRotation(true);
+    for (auto& transceiver : m_peerConnection.getTransceivers()) {
+        if (auto* videoSource = backendFromRTPSender(transceiver->sender()).videoSource())
+            videoSource->setApplyRotation(true);
+    }
 }
 
 bool LibWebRTCPeerConnectionBackend::shouldOfferAllowToReceive(const char* kind) const

Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h (235721 => 235722)


--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h	2018-09-06 04:07:53 UTC (rev 235721)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h	2018-09-06 05:08:43 UTC (rev 235722)
@@ -52,9 +52,6 @@
     LibWebRTCPeerConnectionBackend(RTCPeerConnection&, LibWebRTCProvider&);
     ~LibWebRTCPeerConnectionBackend();
 
-    bool hasAudioSources() const { return m_audioSources.size(); }
-    bool hasVideoSources() const { return m_videoSources.size(); }
-
     void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&);
 
     bool shouldOfferAllowToReceive(const char*) const;
@@ -84,8 +81,6 @@
     friend class LibWebRTCMediaEndpoint;
     friend class LibWebRTCRtpSenderBackend;
     RTCPeerConnection& connection() { return m_peerConnection; }
-    void addAudioSource(Ref<RealtimeOutgoingAudioSource>&&);
-    void addVideoSource(Ref<RealtimeOutgoingVideoSource>&&);
 
     void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
 
@@ -119,8 +114,6 @@
     bool m_isRemoteDescriptionSet { false };
 
     Vector<std::unique_ptr<webrtc::IceCandidateInterface>> m_pendingCandidates;
-    Vector<Ref<RealtimeOutgoingAudioSource>> m_audioSources;
-    Vector<Ref<RealtimeOutgoingVideoSource>> m_videoSources;
     Vector<Ref<RTCRtpReceiver>> m_pendingReceivers;
 };

Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h (235721 => 235722)


--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h	2018-09-06 04:07:53 UTC (rev 235721)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h	2018-09-06 05:08:43 UTC (rev 235722)
@@ -28,6 +28,8 @@
 
 #include "LibWebRTCMacros.h"
 #include "RTCRtpSenderBackend.h"
+#include "RealtimeOutgoingAudioSource.h"
+#include "RealtimeOutgoingVideoSource.h"
 #include <wtf/WeakPtr.h>
 
 #pragma GCC diagnostic push
@@ -53,6 +55,42 @@
     void setRTCSender(rtc::scoped_refptr<webrtc::RtpSenderInterface>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
     webrtc::RtpSenderInterface* rtcSender() { return m_rtcSender.get(); }
 
+    RealtimeOutgoingAudioSource* audioSource()
+    {
+        return WTF::switchOn(m_source,
+            [] (Ref<RealtimeOutgoingAudioSource>& source) { return source.ptr(); },
+            [] (const auto&) -> RealtimeOutgoingAudioSource* { return nullptr; }
+        );
+    }
+
+    RealtimeOutgoingVideoSource* videoSource()
+    {
+        return WTF::switchOn(m_source,
+            [] (Ref<RealtimeOutgoingVideoSource>& source) { return source.ptr(); },
+            [] (const auto&) -> RealtimeOutgoingVideoSource* { return nullptr; }
+        );
+    }
+
+    bool hasNoSource() const
+    {
+        return WTF::switchOn(m_source,
+            [] (const std::nullptr_t&) { return true; },
+            [] (const auto&) { return false; }
+        );
+    }
+
+    void setSource(Ref<RealtimeOutgoingAudioSource>&& source)
+    {
+        ASSERT(hasNoSource());
+        m_source = WTFMove(source);
+    }
+
+    void setSource(Ref<RealtimeOutgoingVideoSource>&& source)
+    {
+        ASSERT(hasNoSource());
+        m_source = WTFMove(source);
+    }
+
 private:
     void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
     RTCRtpParameters getParameters() const final;
@@ -60,6 +98,7 @@
 
     WeakPtr<LibWebRTCPeerConnectionBackend> m_peerConnectionBackend;
     rtc::scoped_refptr<webrtc::RtpSenderInterface> m_rtcSender;
+    Variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSource>, Ref<RealtimeOutgoingVideoSource>> m_source;
 };
 
 } // namespace WebCore

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