webrtc_m130

androidtests

Add support for scaling textures in AndroidVideoCapturer.

2015-11-26 12:41:52 +00:00

java

Android SurfaceViewRenderer: Add resetStatistics() method

2015-11-28 11:15:04 +00:00

javatests

Remove references to libpeerconnection.

2015-10-08 12:41:05 +00:00

objc

Fix target_arch for ios devices

2015-11-24 00:22:32 +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

Reland of Adding the ability to create an RtpSender without a track.

2015-11-25 19:26:08 +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

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

peerconnectionfactory_unittest.cc

Catching more errors when parsing ICE server URLs.

2015-10-06 18:38:33 +00:00

peerconnectionfactory.cc

Added StopAecDump function to PeerConnectionFactory.

2015-10-22 10:25:45 +00:00

peerconnectionfactory.h

Added StopAecDump function to PeerConnectionFactory.

2015-10-22 10:25:45 +00:00

peerconnectionfactoryproxy.h

Added StopAecDump function to PeerConnectionFactory.

2015-10-22 10:25:45 +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

Reland of Adding the ability to create an RtpSender without a track.

2015-11-25 19:26:08 +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

Removed dummy "mediastreamsignaling.h"

2015-11-19 20:01:10 +00:00

webrtcsession.cc

Reland of Adding the ability to create an RtpSender without a track.

2015-11-25 19:26:08 +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