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d82eee0675cf971cdc5f02340b5799bb303449e3
platform-external-webrtc/webrtc/api
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zhihuang d82eee0675 Log how often DTLS negotiation failed because of incompatible ciphersuites. 
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.

BUG=webrtc:5959

Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
2016-08-26 18:25:09 +00:00
..
android
Revert of Delete method cricket::VideoFrame::Copy. (patchset #3 id:210001 of https://codereview.webrtc.org/2275243002/ )
2016-08-25 14:08:19 +00:00
androidtests
Convert PeerConnectionTest to use the new capture APIs.
2016-08-12 07:06:22 +00:00
test
Adding test for unordered, fragmented SCTP message delivery.
2016-08-18 18:40:45 +00:00
androidvideocapturer.cc
Reland of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2106333005/ )
2016-07-01 12:10:59 +00:00
androidvideocapturer.h
AndroidVideoCapturer: Remove unused member variable
2016-06-30 08:05:46 +00:00
androidvideotracksource.cc
Add cricket::VideoFrame::transport_frame_id() and set it to RTP timestamp.
2016-08-01 20:36:04 +00:00
androidvideotracksource.h
This implementation greatly simplifies Android video capturing stack. The old
2016-07-20 14:13:20 +00:00
api_java.gyp
Reland of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2106333005/ )
2016-07-01 12:10:59 +00:00
api_tests.gyp
Moving mock classes around so that they may be reused in other unittests
2016-08-16 08:19:48 +00:00
api.gyp
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +00:00
audiotrack.cc
Propagate MediaStreamSource state to video tracks the same way as audio.
2016-03-23 07:34:01 +00:00
audiotrack.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
BUILD.gn
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +00:00
datachannel_unittest.cc
Fix for data channels perpetually stuck in "closing" state.
2016-08-12 17:10:42 +00:00
datachannel.cc
Fixing off-by-one error with max SCTP id.
2016-08-24 20:15:35 +00:00
datachannel.h
Fix for data channels perpetually stuck in "closing" state.
2016-08-12 17:10:42 +00:00
datachannelinterface.h
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
dtmfsender_unittest.cc
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
dtmfsender.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
dtmfsender.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
dtmfsenderinterface.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
fakemediacontroller.h
Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection.
2016-03-01 20:42:08 +00:00
fakemetricsobserver.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
fakemetricsobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
jsep.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepicecandidate.cc
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepicecandidate.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepsessiondescription_unittest.cc
Replace scoped_ptr with unique_ptr in webrtc/api/
2016-04-27 13:47:40 +00:00
jsepsessiondescription.cc
Replace scoped_ptr with unique_ptr in webrtc/api/
2016-04-27 13:47:40 +00:00
jsepsessiondescription.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
localaudiosource_unittest.cc
This CL provides interfaces that do not use constraints for
2016-03-04 10:51:44 +00:00
localaudiosource.cc
This CL adds activation logic of the new APM level control
2016-06-30 07:02:41 +00:00
localaudiosource.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
mediaconstraintsinterface_unittest.cc
Embed a cricket::MediaConfig in RTCConfiguration.
2016-04-12 06:25:34 +00:00
mediaconstraintsinterface.cc
This CL adds activation logic of the new APM level control
2016-06-30 07:02:41 +00:00
mediaconstraintsinterface.h
This CL adds activation logic of the new APM level control
2016-06-30 07:02:41 +00:00
mediacontroller.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
mediacontroller.h
Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection.
2016-03-01 20:42:08 +00:00
mediastream_unittest.cc
Replace scoped_ptr with unique_ptr in webrtc/api/
2016-04-27 13:47:40 +00:00
mediastream.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastream.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
mediastreaminterface.h
Fix VideoFrame inclusion in mediastreaminterface.h
2016-07-14 14:14:31 +00:00
mediastreamobserver.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamproxy.h
Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
2016-04-15 10:49:13 +00:00
mediastreamtrack.h
Removed MediaStreamTrackInterface::set_state
2016-03-24 10:16:23 +00:00
mediastreamtrackproxy.h
Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
2016-04-15 10:49:13 +00:00
notifier.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
OWNERS
This implementation greatly simplifies Android video capturing stack. The old
2016-07-20 14:13:20 +00:00
peerconnection_unittest.cc
Adding test for unordered, fragmented SCTP message delivery.
2016-08-18 18:40:45 +00:00
peerconnection_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
peerconnection_unittests.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
peerconnection.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
peerconnection.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
peerconnectionendtoend_unittest.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
peerconnectionfactory_unittest.cc
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00
peerconnectionfactory.cc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
peerconnectionfactory.h
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
peerconnectionfactoryproxy.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
peerconnectioninterface_unittest.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
peerconnectioninterface.h
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
peerconnectionproxy.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
proxy_unittest.cc
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
proxy.h
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
quicdatachannel_unittest.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.h
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatatransport_unittest.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.h
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
remoteaudiosource.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
remoteaudiosource.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtcstats.h
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +00:00
rtcstatsreport.h
RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 08:33:19 +00:00
rtpparameters.h
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
rtpreceiver.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpreceiver.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpreceiverinterface.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsender.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsender.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsenderinterface.h
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
rtpsenderreceiver_unittest.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
sctputils_unittest.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
sctputils.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
sctputils.h
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
statscollector_unittest.cc
Moving mock classes around so that they may be reused in other unittests
2016-08-16 08:19:48 +00:00
statscollector.cc
Refactor certificate stats collection, added SSLCertificateStats.
2016-08-25 10:52:46 +00:00
statscollector.h
Refactor certificate stats collection, added SSLCertificateStats.
2016-08-25 10:52:46 +00:00
statstypes.cc
Add Stats to Stun ping.
2016-06-02 00:09:24 +00:00
statstypes.h
Add Stats to Stun ping.
2016-06-02 00:09:24 +00:00
streamcollection.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
umametrics.h
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
videocapturertracksource_unittest.cc
Replace scoped_ptr with unique_ptr in webrtc/api/
2016-04-27 13:47:40 +00:00
videocapturertracksource.cc
Fix some sign-compare warnings in webrtc/api.
2016-06-20 14:04:19 +00:00
videocapturertracksource.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
videosourceproxy.h
Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
2016-04-15 10:49:13 +00:00
videotrack_unittest.cc
Replace scoped_ptr with unique_ptr in webrtc/api/
2016-04-27 13:47:40 +00:00
videotrack.cc
Extended proxy abstraction, to call certain methods to the worker thread.
2016-04-07 14:46:00 +00:00
videotrack.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
videotracksource.cc
Extended proxy abstraction, to call certain methods to the worker thread.
2016-04-07 14:46:00 +00:00
videotracksource.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
webrtcsdp_unittest.cc
RFC 3984 sprop-parameter-sets SDP unit test
2016-06-03 08:22:47 +00:00
webrtcsdp.cc
Fixing off-by-one error with max SCTP id.
2016-08-24 20:15:35 +00:00
webrtcsdp.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
webrtcsession_unittest.cc
Fixing inconsistency with behavior of ClearGettingPorts.
2016-08-24 22:15:07 +00:00
webrtcsession.cc
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
webrtcsession.h
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
webrtcsessiondescriptionfactory.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
webrtcsessiondescriptionfactory.h
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00