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[WebKit-https.git] / Source / ThirdParty / libwebrtc / ChangeLog
1 2017-01-30  Youenn Fablet  <youennf@gmail.com>
2
3         [WebRTC] Upload a diff of WebKit libwebrtc code and original libwebrtc code
4         https://bugs.webkit.org/show_bug.cgi?id=167573
5
6         Reviewed by Alex Christensen.
7
8         * WebKit/patch-libwebrtc: Added.
9
10 2017-01-27  Dan Bernstein  <mitz@apple.com>
11
12         Ignore Xcode’s project.xcworkspace and userdata directories in this new project like we do
13         in other projects.
14
15         * libwebrtc.xcodeproj: Added property svn:ignore.
16
17 2017-01-24  Youenn Fablet  <youenn@apple.com>
18
19         [WebRTC] Use HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE for libwebrtc
20         https://bugs.webkit.org/show_bug.cgi?id=167353
21
22         Reviewed by Alex Christensen.
23
24         * CMakeLists.txt:
25
26 2017-01-23  Youenn Fablet  <youenn@apple.com>
27
28         [WebRTC] Filter libwebrtc link flags
29         https://bugs.webkit.org/show_bug.cgi?id=167287
30
31         Reviewed by Alex Christensen.
32
33         * CMakeLists.txt:
34
35 2017-01-23  Youenn Fablet  <youennf@gmail.com>
36
37         [WebRTC] Make VP8 optional in libwebrtc
38         https://bugs.webkit.org/show_bug.cgi?id=167257
39
40         Reviewed by Darin Adler.
41
42         Reusing strategy used to have VP9 optional for VP8 codec.
43
44         * CMakeLists.txt: Updated tocompile and link vp8_noop.cc
45         * Source/webrtc/media/engine/webrtcvideoengine2.cc:
46         * Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h:
47         * Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc:
48         * Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc: Added.
49         * Source/webrtc/video/video_encoder.cc:
50
51 2017-01-20  Youenn Fablet  <youennf@gmail.com>
52
53         [WebRTC] Update build system to make G711 optional in libwebrtc
54         https://bugs.webkit.org/show_bug.cgi?id=167256
55
56         Reviewed by Alex Christensen.
57
58         * CMakeLists.txt: Updating to add compilation of generic pcm encoder functions.
59
60 2017-01-20  Youenn Fablet  <youennf@gmail.com>
61
62         [WebRTC] Update libwertc AudioRtpSender::SetAudioSend
63         https://bugs.webkit.org/show_bug.cgi?id=167243
64
65         Reviewed by Alex Christensen.
66
67         Introducing  WEBRTC_WEBKIT_BUILD macro to match existing WEBRTC_CHROMIUM_BUILD.
68         WEBRTC_WEBKIT_BUILD is defined by current WebKit libwebrtc build system.
69
70         * Source/webrtc/api/rtpsender.cc:
71
72 2017-01-20  Youenn Fablet  <youennf@gmail.com>
73
74         [WebRTC] libwebrtc NO_RETURN is conflicting with WebKit one
75         https://bugs.webkit.org/show_bug.cgi?id=167244
76
77         Reviewed by Alex Christensen.
78
79         * Source/webrtc/typedefs.h: Defining NO_RETURN only if not already defined.
80
81 2017-01-20  Youenn Fablet  <youenn@apple.com>
82
83         [WebRTC] libwebrtc headers are incompatible with WebKit compilation flags
84         https://bugs.webkit.org/show_bug.cgi?id=167242
85
86         Reviewed by Alex Christensen.
87
88         WebKit is enforcing -Wunused-parameter and -Wunused-variable which conflict with some included libwertc headers.
89         Removed unused parameter names for inlined functions.
90
91         * Source/webrtc/api/jsep.h:
92         (webrtc::SessionDescriptionInterface::RemoveCandidates):
93         * Source/webrtc/api/mediastreaminterface.h:
94         (webrtc::AudioSourceInterface::SetVolume):
95         (webrtc::AudioSourceInterface::RegisterAudioObserver):
96         (webrtc::AudioSourceInterface::UnregisterAudioObserver):
97         (webrtc::AudioSourceInterface::AddSink):
98         (webrtc::AudioSourceInterface::RemoveSink):
99         (webrtc::AudioTrackInterface::GetSignalLevel):
100         * Source/webrtc/api/peerconnectionfactory.h:
101         * Source/webrtc/api/peerconnectioninterface.h:
102         (webrtc::MetricsObserverInterface::IncrementEnumCounter):
103         (webrtc::PeerConnectionInterface::AddTrack):
104         (webrtc::PeerConnectionInterface::RemoveTrack):
105         (webrtc::PeerConnectionInterface::CreateSender):
106         (webrtc::PeerConnectionInterface::GetStats):
107         (webrtc::PeerConnectionInterface::CreateOffer):
108         (webrtc::PeerConnectionInterface::CreateAnswer):
109         (webrtc::PeerConnectionInterface::UpdateIce):
110         (webrtc::PeerConnectionInterface::SetConfiguration):
111         (webrtc::PeerConnectionInterface::RemoveIceCandidates):
112         (webrtc::PeerConnectionInterface::StartRtcEventLog):
113         (webrtc::PeerConnectionObserver::OnAddStream):
114         (webrtc::PeerConnectionObserver::OnRemoveStream):
115         (webrtc::PeerConnectionObserver::OnDataChannel):
116         (webrtc::PeerConnectionObserver::OnIceCandidatesRemoved):
117         (webrtc::PeerConnectionObserver::OnIceConnectionReceivingChange):
118         * Source/webrtc/api/rtpsender.cc:
119         * Source/webrtc/base/messagehandler.h:
120         (rtc::FunctorMessageHandler::OnMessage):
121         * Source/webrtc/base/sanitizer.h:
122         (rtc_AsanPoison):
123         (rtc_AsanUnpoison):
124         (rtc_MsanMarkUninitialized):
125         (rtc_MsanCheckInitialized):
126         * Source/webrtc/base/stream.h:
127         (rtc::StreamInterface::ConsumeReadData):
128         (rtc::StreamInterface::ConsumeWriteBuffer):
129         * Source/webrtc/media/base/mediachannel.h:
130         (cricket::DataMediaChannel::GetStats):
131         (cricket::DataMediaChannel::OnNetworkRouteChanged):
132         * Source/webrtc/media/engine/webrtcvideodecoderfactory.h:
133         (cricket::WebRtcVideoDecoderFactory::CreateVideoDecoderWithParams):
134         * Source/webrtc/media/engine/webrtcvideoencoderfactory.h:
135         (cricket::WebRtcVideoEncoderFactory::VideoCodec::VideoCodec):
136         (cricket::WebRtcVideoEncoderFactory::EncoderTypeHasInternalSource):
137         * Source/webrtc/media/engine/webrtcvideoengine2.cc:
138         * Source/webrtc/modules/include/module.h:
139         (webrtc::Module::ProcessThreadAttached):
140         * Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc:
141         * Source/webrtc/p2p/base/port.h:
142         (cricket::Port::HandleIncomingPacket):
143         (cricket::Port::HandleConnectionDestroyed):
144         (cricket::Connection::set_receiving_timeout):
145         * Source/webrtc/p2p/base/stun.h:
146         (cricket::StunAttribute::SetOwner):
147         * Source/webrtc/p2p/base/stunrequest.h:
148         (cricket::StunRequest::Prepare):
149         (cricket::StunRequest::OnResponse):
150         (cricket::StunRequest::OnErrorResponse):
151         * Source/webrtc/p2p/base/transport.h:
152         (cricket::Transport::SetLocalCertificate):
153         (cricket::Transport::GetLocalCertificate):
154         (cricket::Transport::GetSslRole):
155         (cricket::Transport::SetSslMaxProtocolVersion):
156         * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc:
157         * Source/webrtc/typedefs.h:
158
159 2017-01-20  Youenn Fablet  <youennf@gmail.com>
160
161         [WebRTC] Update libwertc AudioRtpSender::SetAudioSend
162         https://bugs.webkit.org/show_bug.cgi?id=167243
163
164         Reviewed by Alex Christensen.
165
166         Introducing  WEBRTC_WEBKIT_BUILD macro to match existing WEBRTC_CHROMIUM_BUILD.
167         WEBRTC_WEBKIT_BUILD is defined by current WebKit libwebrtc build system.
168
169         * Source/webrtc/api/rtpsender.cc:
170
171 2017-01-20  Youenn Fablet  <youennf@gmail.com>
172
173         [WebRTC] libwebrtc H.264 codec is using VTB only for IOS
174         https://bugs.webkit.org/show_bug.cgi?id=167245
175
176         Reviewed by Alex Christensen.
177
178         * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc: Removing WEBRTC_IOS flag.
179
180 2017-01-19  Youenn Fablet  <youenn@apple.com>
181
182         [WebRTC] Upload libwebrtc code base
183         https://bugs.webkit.org/show_bug.cgi?id=167205
184
185         Reviewed by Alex Christensen and Jon Lee.
186
187         Add initial libwebrtc source from branch 56. Here's how to get what we committed:
188         git clone https://chromium.googlesource.com/external/webrtc.git && cd webrtc && git checkout 7bf536976366443ea59153ff3d22da0ec32badc1