admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/talk/app/webrtc
History
deadbeef 1387149ad1 Adding reduced size RTCP configuration down to the video stream level. 
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.

BUG=webrtc:4868

Review URL: https://codereview.webrtc.org/1418123003

Cr-Commit-Position: refs/heads/master@{#10958}
2015-12-09 20:37:59 +00:00
..
androidtests
Refactor EglBase configuration.
2015-12-07 09:17:22 +00:00
java
Don't call SetPreviewFormat if capturing to textures.
2015-12-08 20:03:07 +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
Allow remote fingerprint update during a call
2015-12-05 18:00:04 +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
Use @webrtc.org addresses for OWNERS.
2015-12-07 18:53:25 +00:00
peerconnection_unittest.cc
Roll gtest-parallel.
2015-12-07 15:45:08 +00:00
peerconnection.cc
Fixing issue with default stream upon setting 2nd remote description.
2015-12-09 01:13:53 +00:00
peerconnection.h
Fixing issue with default stream upon setting 2nd remote description.
2015-12-09 01:13:53 +00:00
peerconnectionendtoend_unittest.cc
Free SCTP data channels asynchronously in PeerConnection.
2015-12-07 23:32:31 +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
Fixing issue with default stream upon setting 2nd remote description.
2015-12-09 01:13:53 +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
remoteaudiotrack.cc
Add empty placeholder files for remote audio tracks.
2015-12-09 17:42:03 +00:00
remoteaudiotrack.h
Add empty placeholder files for remote audio tracks.
2015-12-09 17:42:03 +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
Adding reduced size RTCP configuration down to the video stream level.
2015-12-09 20:37:59 +00:00
webrtcsdp.cc
Adding reduced size RTCP configuration down to the video stream level.
2015-12-09 20:37:59 +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
Add tracing to public PeerConnection methods.
2015-12-08 21:15:26 +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