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381b4217cb36f434c56e399a852a0a15522a9596
platform-external-webrtc/talk/app/webrtc
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Honghai Zhang 381b4217cb Ping backup connection at a slower rate 
and make it configurable from the app.
Changed the decision on whether a connection is pingable:
1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection.
2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate.
Note the default behavior is the same as before.

Also cached the channel state since we are accessing it more often.
BUG=webrtc:5034
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1455033004 .

Cr-Commit-Position: refs/heads/master@{#10900}
2015-12-04 20:24:10 +00:00
..
androidtests
This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase.
2015-12-02 09:07:22 +00:00
java
Ping backup connection at a slower rate
2015-12-04 20:24:10 +00:00
javatests
Remove references to libpeerconnection.
2015-10-08 12:41:05 +00:00
objc
Ping backup connection at a slower rate
2015-12-04 20:24:10 +00:00
objctests
Add OWNERS for ObjC dirs.
2015-10-01 00:33:36 +00:00
test
Removed dummy "mediastreamsignaling.h"
2015-11-19 20:01:10 +00:00
androidvideocapturer.cc
Add support for scaling textures in AndroidVideoCapturer.
2015-11-26 12:41:52 +00:00
androidvideocapturer.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
audiotrack.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
audiotrack.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
datachannel_unittest.cc
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
datachannel.cc
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
datachannel.h
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
datachannelinterface.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
dtlsidentitystore_unittest.cc
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
dtlsidentitystore.cc
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
dtlsidentitystore.h
New DtlsIdentityStoreInterface::RequestIdentity added that takes rtc::KeyParams. The old RequestIdentity still exists that take rtc::KeyType.
2015-10-21 08:44:27 +00:00
dtmfsender_unittest.cc
Fix: RefCountInterface: Make AddRef() and Release() const
2015-10-12 14:10:50 +00:00
dtmfsender.cc
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
dtmfsender.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
dtmfsenderinterface.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
fakemediacontroller.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
fakemetricsobserver.cc
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
fakemetricsobserver.h
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
fakeportallocatorfactory.h
Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
2015-11-11 20:55:18 +00:00
jsep.h
Remove deprecated functions.
2015-07-16 20:43:27 +00:00
jsepicecandidate.cc
Remove deprecated functions.
2015-07-16 20:43:27 +00:00
jsepicecandidate.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
jsepsessiondescription_unittest.cc
Remove GICE (again).
2015-09-18 01:55:03 +00:00
jsepsessiondescription.cc
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
jsepsessiondescription.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
localaudiosource_unittest.cc
Rename Maybe to Optional
2015-11-10 21:34:32 +00:00
localaudiosource.cc
Rename Maybe to Optional
2015-11-10 21:34:32 +00:00
localaudiosource.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediaconstraintsinterface.cc
Remove simulcast bitrate modes.
2015-10-16 19:49:47 +00:00
mediaconstraintsinterface.h
Remove simulcast bitrate modes.
2015-10-16 19:49:47 +00:00
mediacontroller.cc
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
mediacontroller.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
mediastream_unittest.cc
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediastream.cc
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediastream.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
mediastreamhandler.cc
Exposing RtpSenders and RtpReceivers from PeerConnection.
2015-09-28 23:54:02 +00:00
mediastreamhandler.h
Exposing RtpSenders and RtpReceivers from PeerConnection.
2015-09-28 23:54:02 +00:00
mediastreaminterface.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
mediastreamprovider.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
mediastreamproxy.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediastreamtrack.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediastreamtrackproxy.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
notifier.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
OWNERS
Cleanup OWNERS of talk/app/webrtc.
2015-10-13 16:53:37 +00:00
peerconnection_unittest.cc
Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.
2015-12-04 09:37:10 +00:00
peerconnection.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
peerconnection.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
peerconnectionendtoend_unittest.cc
Disable PeerConnectionEndToEndTest.Call on Mac.
2015-12-02 21:53:39 +00:00
peerconnectionfactory_unittest.cc
Catching more errors when parsing ICE server URLs.
2015-10-06 18:38:33 +00:00
peerconnectionfactory.cc
Adding CreatePeerConnection method that uses new PC Initialize method.
2015-12-01 23:10:17 +00:00
peerconnectionfactory.h
Adding CreatePeerConnection method that uses new PC Initialize method.
2015-12-01 23:10:17 +00:00
peerconnectionfactoryproxy.h
Adding CreatePeerConnection method that uses new PC Initialize method.
2015-12-01 23:10:17 +00:00
peerconnectioninterface_unittest.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
peerconnectioninterface.h
Ping backup connection at a slower rate
2015-12-04 20:24:10 +00:00
peerconnectionproxy.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
portallocatorfactory.cc
Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
2015-11-11 20:55:18 +00:00
portallocatorfactory.h
Makes libjingle_peerconnection_android_unittest run on networkless devices.
2015-02-12 09:24:25 +00:00
proxy_unittest.cc
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
proxy.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
remoteaudiosource.cc
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
remoteaudiosource.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
remotevideocapturer_unittest.cc
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
remotevideocapturer.cc
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
remotevideocapturer.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
rtpreceiver.cc
Remove MediaChannel::SetRemoteRenderer().
2015-10-09 15:55:54 +00:00
rtpreceiver.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
rtpreceiverinterface.h
Exposing RtpSenders and RtpReceivers from PeerConnection.
2015-09-28 23:54:02 +00:00
rtpsender.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
rtpsender.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
rtpsenderinterface.h
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
rtpsenderreceiver_unittest.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
sctputils_unittest.cc
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
sctputils.cc
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
sctputils.h
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00
statscollector_unittest.cc
Removed dummy "mediastreamsignaling.h"
2015-11-19 20:01:10 +00:00
statscollector.cc
Don't overwrite audio stats when they're not available.
2015-11-28 01:27:40 +00:00
statscollector.h
Removed dummy "mediastreamsignaling.h"
2015-11-19 20:01:10 +00:00
statstypes.cc
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
statstypes.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
streamcollection.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
umametrics.h
Reland Change WebRTC SslCipher to be exposed as number only
2015-10-01 04:49:02 +00:00
videosource_unittest.cc
Rename Maybe to Optional
2015-11-10 21:34:32 +00:00
videosource.cc
Temporarily disable VERIFY while bug is investigated.
2015-11-18 17:55:00 +00:00
videosource.h
Add VideoSource::Stop and Restart methods.
2015-02-17 13:54:42 +00:00
videosourceinterface.h
Revert "Add default implementation to VideoSourceInterface."
2015-03-02 11:33:41 +00:00
videosourceproxy.h
Add VideoSource::Stop and Restart methods.
2015-02-17 13:54:42 +00:00
videotrack_unittest.cc
Remove cricket::VideoFrame::Set/GetElapsedTime()
2015-10-01 10:03:05 +00:00
videotrack.cc
Reland of Adding the ability to create an RtpSender without a track.
2015-11-25 19:26:08 +00:00
videotrack.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
videotrackrenderers.cc
Remove VideoRendererInterface::CanApplyRotation()
2015-09-09 19:26:19 +00:00
videotrackrenderers.h
Remove VideoRendererInterface::CanApplyRotation()
2015-09-09 19:26:19 +00:00
webrtcsdp_unittest.cc
Fixing some issues with the direction attribute of m-lines in offers.
2015-10-22 20:14:51 +00:00
webrtcsdp.cc
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
webrtcsdp.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
webrtcsession_unittest.cc
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
2015-12-02 20:35:14 +00:00
webrtcsession.cc
Ping backup connection at a slower rate
2015-12-04 20:24:10 +00:00
webrtcsession.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
webrtcsessiondescriptionfactory.cc
Merging BaseSession code into WebRtcSession.
2015-10-14 22:02:50 +00:00
webrtcsessiondescriptionfactory.h
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
2015-10-14 18:33:20 +00:00