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webrtc_m130/talk/app/webrtc
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deadbeef b5cb19b37c Fixing direction attribute in answer for non-RTP protocols. 
"non-RTP protocols" refers to SCTP data channels. Because
there are no streams for SCTP data channels, the answer was being
set to RECVONLY.

BUG=webrtc:5228

Review URL: https://codereview.webrtc.org/1473013002

Cr-Commit-Position: refs/heads/master@{#10762}
2015-11-24 00:39:18 +00:00
..
androidtests
This cl add support to encode from textures to MediaCodecVideoEncoder.
2015-11-20 09:31:32 +00:00
java
Android ThreadUtils: Make the class public for access outside org.webrtc
2015-11-23 19:13:36 +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
Chrome has now been updated.
2015-11-19 20:02:28 +00:00
androidvideocapturer.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
audiotrack.cc
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
mediastreamprovider.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Fixing direction attribute in answer for non-RTP protocols.
2015-11-24 00:39:18 +00:00
peerconnection.cc
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
peerconnection.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
peerconnectioninterface.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
peerconnectionproxy.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
rtpsender.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
rtpsenderinterface.h
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +00:00
rtpsenderreceiver_unittest.cc
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Reland Convert internal representation of Srtp cryptos from string to int
2015-11-19 03:42:00 +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
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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