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8f741e953c1d712cbf79a96bf2f9afb73542154f
platform-external-webrtc/webrtc/api
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sakal 8f741e953c Change Camera1Enumerator to create a Camera1Capturer instead of VideoCapturerAndroid. 
BUG=webrtc:6148

Review-Url: https://codereview.webrtc.org/2356073003
Cr-Commit-Position: refs/heads/master@{#14478}
2016-10-03 15:06:05 +00:00
..
android
Change Camera1Enumerator to create a Camera1Capturer instead of VideoCapturerAndroid.
2016-10-03 15:06:05 +00:00
androidtests
Format all Java in WebRTC.
2016-09-29 11:12:51 +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
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
androidvideotracksource.cc
New class AdaptedVideoTrackSource.
2016-09-22 08:26:06 +00:00
androidvideotracksource.h
New class AdaptedVideoTrackSource.
2016-09-22 08:26:06 +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
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
GN: Fix incorrect include_dir for libjingle_peerconnection_jni target
2016-09-28 07:47:59 +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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +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
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +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
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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +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
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
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
rtpsenderreceiver_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +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
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
statscollector.cc
Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere
2016-10-03 12:04:25 +00:00
statscollector.h
Refactor certificate stats collection, added SSLCertificateStats.
2016-08-25 10:52:46 +00:00
statstypes.cc
Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere
2016-10-03 12:04:25 +00:00
statstypes.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +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
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +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
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
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
webrtcsdp_unittest.cc
Fixing heap read overflow when "sctp-port" is in a video description.
2016-09-28 17:04:41 +00:00
webrtcsdp.cc
Fixing heap read overflow when "sctp-port" is in a video description.
2016-09-28 17:04:41 +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
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
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