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webrtc_m130/webrtc/api
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hbos 664efbd048 Removed api/rtcstats[report].h pseudonyms of api/stats/rtcstats[report].h 
These are no longer used in Chromium, so deleting them will not break any
third party project.

BUG=chromium:627816
NOTRY=True

Review-Url: https://codereview.webrtc.org/2352993002
Cr-Commit-Position: refs/heads/master@{#14309}
2016-09-20 14:10:19 +00:00
..
android
Enable auto-focus on Camera2.
2016-09-19 13:01:59 +00:00
androidtests
Change onCameraOpening to take camera name as a parameter instead of camera id.
2016-09-14 12:36:26 +00:00
call
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
stats
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
test
RTCStatsCollector and RTCPeerConnectionStats added.
2016-08-30 21:04:40 +00:00
androidvideotracksource.cc
Reland of Optimize Android NV12 capture (patchset #1 id:1 of https://codereview.webrtc.org/2327893002/ )
2016-09-17 09:39:11 +00:00
androidvideotracksource.h
Reland of Optimize Android NV12 capture (patchset #1 id:1 of https://codereview.webrtc.org/2327893002/ )
2016-09-17 09:39:11 +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.gyp
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
audiotrack.cc
…
audiotrack.h
…
BUILD.gn
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +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
…
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
dtmfsender_unittest.cc
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dtmfsender.cc
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dtmfsender.h
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dtmfsenderinterface.h
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fakemediacontroller.h
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fakemetricsobserver.cc
…
fakemetricsobserver.h
…
jsep.h
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jsepicecandidate.cc
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jsepicecandidate.h
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jsepsessiondescription_unittest.cc
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jsepsessiondescription.cc
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jsepsessiondescription.h
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localaudiosource_unittest.cc
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localaudiosource.cc
This CL adds activation logic of the new APM level control
2016-06-30 07:02:41 +00:00
localaudiosource.h
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mediaconstraintsinterface_unittest.cc
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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
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mediacontroller.h
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mediastream_unittest.cc
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mediastream.cc
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mediastream.h
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mediastreaminterface.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
mediastreamobserver.cc
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mediastreamobserver.h
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mediastreamproxy.h
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mediastreamtrack.h
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mediastreamtrackproxy.h
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notifier.h
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OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
peerconnection_unittest.cc
Adding an end-to-end connection time test.
2016-09-16 22:07:58 +00:00
peerconnection.cc
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
peerconnection.h
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +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 a switch to redetermine role when ICE restarts.
2016-08-31 05:07:56 +00:00
peerconnectionfactory.h
Add a switch to redetermine role when ICE restarts.
2016-08-31 05:07:56 +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
Add a switch to redetermine role when ICE restarts.
2016-08-31 05:07:56 +00:00
peerconnectioninterface.h
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
peerconnectionproxy.h
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
proxy_unittest.cc
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proxy.h
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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
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
rtcstatscollector_unittest.cc
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
rtcstatscollector.cc
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +00:00
rtcstatscollector.h
PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added.
2016-09-16 06:33:04 +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
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rtpsenderreceiver_unittest.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
sctputils_unittest.cc
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sctputils.cc
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sctputils.h
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statscollector_unittest.cc
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
statscollector.cc
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
statscollector.h
Refactor certificate stats collection, added SSLCertificateStats.
2016-08-25 10:52:46 +00:00
statstypes.cc
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
statstypes.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
streamcollection.h
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umametrics.h
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
videocapturertracksource_unittest.cc
Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
2016-09-14 19:03:21 +00:00
videocapturertracksource.cc
Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
2016-09-14 19:03:21 +00:00
videocapturertracksource.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
videosourceproxy.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
videotrack_unittest.cc
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videotrack.cc
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videotrack.h
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videotracksource.cc
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videotracksource.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
webrtcsdp_unittest.cc
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webrtcsdp.cc
Fixing off-by-one error with max SCTP id.
2016-08-24 20:15:35 +00:00
webrtcsdp.h
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webrtcsession_unittest.cc
Fixing inconsistency with behavior of ClearGettingPorts.
2016-08-24 22:15:07 +00:00
webrtcsession.cc
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
webrtcsession.h
Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 18:25:09 +00:00
webrtcsessiondescriptionfactory.cc
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webrtcsessiondescriptionfactory.h
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00