Attachment 348956 Details for Bug 189310 – Patch

[patch] Patch

bug-189310-20180905131212.patch (text/plain), 10.88 KB, created by youenn fablet on 2018-09-05 13:12:13 PDT

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Creator: youenn fablet

Created: 2018-09-05 13:12:13 PDT

Size: 10.88 KB

patch

obsolete

>Subversion Revision: 235683
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index f33370596ced8fd22fd60a3f8577e039fbfc7045..febcf8981b95bc667f848620cb582d8f679a6f4b 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,26 @@
>+2018-09-05  Youenn Fablet  <youenn@apple.com>
>+
>+        Move ownership of outgoing source to RTCRtpSender backend
>+        https://bugs.webkit.org/show_bug.cgi?id=189310
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        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  Woodrow Wang  <woodrow_wang@apple.com>
> 
>         Added runtime feature flag for web API statistics
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index 98cbee5c5a3450f260d6260eb7e4199e4848af63..efa19adc72f4189ebb8940482601a5e6a3f82c1d 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -214,9 +214,9 @@ bool LibWebRTCMediaEndpoint::addTrack(LibWebRTCRtpSenderBackend& sender, MediaSt
> 
>     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 @@ bool LibWebRTCMediaEndpoint::addTrack(LibWebRTCRtpSenderBackend& sender, MediaSt
>     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;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>index 6205a02c4254bdebbda9fa3b1985e2946fefda46..ebecb50d4f56f06a3b49c8e8a4c81a45a8ab01a1 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>@@ -174,15 +174,7 @@ void LibWebRTCPeerConnectionBackend::doCreateAnswer(RTCAnswerOptions&&)
> 
> 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 @@ void LibWebRTCPeerConnectionBackend::doAddIceCandidate(RTCIceCandidate& candidat
>     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() });
>@@ -402,23 +384,10 @@ template<typename Source>
> static inline bool updateTrackSource(Source& source, MediaStreamTrack* track)
> {
>     if (!track) {
>-        source->stop();
>+        source.stop();
>         return true;
>     }
>-    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;
>+    return source.setSource(track->privateTrack());
> }
> 
> void LibWebRTCPeerConnectionBackend::replaceTrack(RTCRtpSender& sender, RefPtr<MediaStreamTrack>&& track, DOMPromiseDeferred<void>&& promise)
>@@ -433,13 +402,13 @@ void LibWebRTCPeerConnectionBackend::replaceTrack(RTCRtpSender& sender, RefPtr<M
>             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;
>             }
>@@ -475,8 +444,10 @@ void LibWebRTCPeerConnectionBackend::enqueueReplaceTrackTask(RTCRtpSender& sende
> 
> 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
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>index 64e609c260504f02f790366be0a043b8a1b02c04..5f5e02936e33d437df4450e1491eba911e93f3ee 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>@@ -52,9 +52,6 @@ public:
>     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;
>@@ -85,8 +82,6 @@ private:
>     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 @@ private:
>     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;
> };
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>index 24b1b1bc7d466481d81f74665d9c77625201718f..686fa88f3ed516b0523b2319f01a6f9d85cccd68 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderBackend.h
>@@ -28,6 +28,8 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RTCRtpSenderBackend.h"
>+#include "RealtimeOutgoingAudioSource.h"
>+#include "RealtimeOutgoingVideoSource.h"
> #include <wtf/WeakPtr.h>
> 
> #pragma GCC diagnostic push
>@@ -53,12 +55,49 @@ public:
>     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;
> 
>     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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Attachments on bug 189310: 348948 | 348956 | 348966 | 349000