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webrtc_m130/talk/app/webrtc
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kjellander 2f042f26a3 Roll chromium_revision 1b6c421..db567a8 (365999:366304) 
I had to disable some Dtls12Both tests failing under MSan (see bug).
Notice those errors started happening in the range of
https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2
while this CL brings in an even newer BoringSSL (that still has the same problem).

Change log: 1b6c421..db567a8
Full diff: 1b6c421..db567a8

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb
* src/third_party/libyuv: 1019e45..1ccbf8f
* src/third_party/nss: a676aa0..aee1b12
DEPS diff: 1b6c421..db567a8/DEPS

No update to Clang.

NOTRY=True
BUG=webrtc:5381
TBR=torbjorng@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1533253002

Cr-Commit-Position: refs/heads/master@{#11095}
2015-12-20 20:25:17 +00:00
..
androidtests
Use NV21 instead of YUV12 and clean up.
2015-12-18 08:37:10 +00:00
java
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
javatests
Remove references to libpeerconnection.
2015-10-08 12:41:05 +00:00
objc
Add oldest rotation type option to RTCFileLogger
2015-12-08 21:59:11 +00:00
objctests
Add OWNERS for ObjC dirs.
2015-10-01 00:33:36 +00:00
test
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
androidvideocapturer.cc
This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers.
2015-12-11 08:32:50 +00:00
androidvideocapturer.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
audiotrack.cc
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
audiotrack.h
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
dtlsidentitystore.h
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
mediastreamobserver.cc
Restoring behavior where PeerConnection tracks changes to MediaStreams.
2015-12-16 03:24:50 +00:00
mediastreamobserver.h
Restoring behavior where PeerConnection tracks changes to MediaStreams.
2015-12-16 03:24:50 +00:00
mediastreamprovider.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +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
Use @webrtc.org addresses for OWNERS.
2015-12-07 18:53:25 +00:00
peerconnection_unittest.cc
Roll chromium_revision 1b6c421..db567a8 (365999:366304)
2015-12-20 20:25:17 +00:00
peerconnection.cc
Adding a MediaStream parameter to createSender.
2015-12-19 00:58:51 +00:00
peerconnection.h
Adding a MediaStream parameter to createSender.
2015-12-19 00:58:51 +00:00
peerconnectionendtoend_unittest.cc
Fixing and re-enabling some flaky PeerConnection tests.
2015-12-15 02:28:04 +00:00
peerconnectionfactory_unittest.cc
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
peerconnectionfactory.cc
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
peerconnectionfactory.h
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
peerconnectionfactoryproxy.h
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
peerconnectioninterface_unittest.cc
Adding a MediaStream parameter to createSender.
2015-12-19 00:58:51 +00:00
peerconnectioninterface.h
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
2015-12-19 18:14:18 +00:00
peerconnectionproxy.h
Adding a MediaStream parameter to createSender.
2015-12-19 00:58:51 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
remoteaudiosource.h
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
remoteaudiotrack.cc
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
remoteaudiotrack.h
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
rtpreceiver.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
rtpreceiverinterface.h
Exposing RtpSenders and RtpReceivers from PeerConnection.
2015-09-28 23:54:02 +00:00
rtpsender.cc
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
statscollector.cc
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
statscollector.h
Removed dummy "mediastreamsignaling.h"
2015-11-19 20:01:10 +00:00
statstypes.cc
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
statstypes.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
videosource.cc
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
videosource.h
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
videosourceinterface.h
Revert "Add default implementation to VideoSourceInterface."
2015-03-02 11:33:41 +00:00
videosourceproxy.h
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +00:00
videotrack_unittest.cc
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
2015-12-15 12:27:20 +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
Add ufrag to the ICE candidate signaling.
2015-12-17 02:37:27 +00:00
webrtcsdp.cc
Add ufrag to the ICE candidate signaling.
2015-12-17 02:37:27 +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
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
webrtcsession.cc
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +00:00
webrtcsession.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
webrtcsessiondescriptionfactory.cc
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
2015-12-17 11:04:24 +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