Modified: trunk/Source/WebCore/ChangeLog (266453 => 266454)
--- trunk/Source/WebCore/ChangeLog 2020-09-02 09:31:34 UTC (rev 266453)
+++ trunk/Source/WebCore/ChangeLog 2020-09-02 09:45:29 UTC (rev 266454)
@@ -1,3 +1,23 @@
+2020-09-02 Youenn Fablet <you...@apple.com>
+
+ Safari is not able to hear audio when using WebRTC in multiple tabs
+ https://bugs.webkit.org/show_bug.cgi?id=215270
+ <rdar://problem/66736746>
+
+ Reviewed by Eric Carlson.
+
+ Covered by manually testing that audio continues when two pages in the same process create a peer connection,
+ one plays audio with WebRTC and the second one is closed.
+ Before the patch, the closing page would stop the audio device module, even though the first page would like the module to continue playing.
+ Fix this by creating an audio module per page instead of process wide.
+
+ * platform/mediastream/libwebrtc/LibWebRTCAudioModule.h:
+ * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
+ (WebCore::initializePeerConnectionFactoryAndThreads):
+ (WebCore::LibWebRTCProvider::factory):
+ (WebCore::LibWebRTCProvider::createPeerConnectionFactory):
+ * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
+
2020-09-02 Said Abou-Hallawa <sabouhall...@apple.com>
Unreviewed, reverting r266449.
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h (266453 => 266454)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h 2020-09-02 09:31:34 UTC (rev 266453)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h 2020-09-02 09:45:29 UTC (rev 266454)
@@ -40,7 +40,7 @@
namespace WebCore {
// LibWebRTCAudioModule is pulling streamed data to ensure audio data is passed to the audio track.
-class LibWebRTCAudioModule final : public webrtc::AudioDeviceModule {
+class LibWebRTCAudioModule : public webrtc::AudioDeviceModule {
WTF_MAKE_FAST_ALLOCATED;
public:
LibWebRTCAudioModule();
@@ -52,9 +52,6 @@
return value;
}
- void AddRef() const final { }
- rtc::RefCountReleaseStatus Release() const final { return rtc::RefCountReleaseStatus::kOtherRefsRemained; }
-
// webrtc::AudioDeviceModule API
int32_t StartPlayout() final;
int32_t StopPlayout() final;
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp (266453 => 266454)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp 2020-09-02 09:31:34 UTC (rev 266453)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp 2020-09-02 09:45:29 UTC (rev 266454)
@@ -118,7 +118,6 @@
std::unique_ptr<rtc::Thread> networkThread;
std::unique_ptr<rtc::Thread> signalingThread;
bool networkThreadWithSocketServer { false };
- std::unique_ptr<LibWebRTCAudioModule> audioDeviceModule;
std::unique_ptr<rtc::NetworkManager> networkManager;
std::unique_ptr<BasicPacketSocketFactory> packetSocketFactory;
std::unique_ptr<rtc::RTCCertificateGenerator> certificateGenerator;
@@ -187,8 +186,6 @@
result = factoryAndThreads.signalingThread->Start();
ASSERT(result);
-
- factoryAndThreads.audioDeviceModule = makeUnique<LibWebRTCAudioModule>();
}
static inline PeerConnectionFactoryAndThreads& staticFactoryAndThreads()
@@ -263,14 +260,16 @@
auto& factoryAndThreads = getStaticFactoryAndThreads(m_useNetworkThreadWithSocketServer);
- m_factory = createPeerConnectionFactory(factoryAndThreads.networkThread.get(), factoryAndThreads.networkThread.get(), factoryAndThreads.audioDeviceModule.get());
+ m_factory = createPeerConnectionFactory(factoryAndThreads.networkThread.get(), factoryAndThreads.networkThread.get());
return m_factory;
}
-rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> LibWebRTCProvider::createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread, LibWebRTCAudioModule* audioModule)
+rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> LibWebRTCProvider::createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread)
{
- return webrtc::CreatePeerConnectionFactory(networkThread, signalingThread, signalingThread, audioModule, webrtc::CreateBuiltinAudioEncoderFactory(), webrtc::CreateBuiltinAudioDecoderFactory(), createEncoderFactory(), createDecoderFactory(), nullptr, nullptr);
+ auto audioModule = rtc::scoped_refptr<webrtc::AudioDeviceModule>(new rtc::RefCountedObject<LibWebRTCAudioModule>());
+
+ return webrtc::CreatePeerConnectionFactory(networkThread, signalingThread, signalingThread, WTFMove(audioModule), webrtc::CreateBuiltinAudioEncoderFactory(), webrtc::CreateBuiltinAudioDecoderFactory(), createEncoderFactory(), createDecoderFactory(), nullptr, nullptr);
}
std::unique_ptr<webrtc::VideoDecoderFactory> LibWebRTCProvider::createDecoderFactory()
Modified: trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h (266453 => 266454)
--- trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h 2020-09-02 09:31:34 UTC (rev 266453)
+++ trunk/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h 2020-09-02 09:45:29 UTC (rev 266454)
@@ -136,7 +136,7 @@
rtc::scoped_refptr<webrtc::PeerConnectionInterface> createPeerConnection(webrtc::PeerConnectionObserver&, rtc::NetworkManager&, rtc::PacketSocketFactory&, webrtc::PeerConnectionInterface::RTCConfiguration&&, std::unique_ptr<webrtc::AsyncResolverFactory>&&);
- rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread, LibWebRTCAudioModule*);
+ rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> createPeerConnectionFactory(rtc::Thread* networkThread, rtc::Thread* signalingThread);
virtual std::unique_ptr<webrtc::VideoDecoderFactory> createDecoderFactory();
virtual std::unique_ptr<webrtc::VideoEncoderFactory> createEncoderFactory();
