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ac9d92ccbe2b29590c53f702e11dc625820480d5
platform-external-webrtc/talk/app/webrtc
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deadbeef ac9d92ccbe Adding the ability to create an RtpSender without a track. 
This CL also changes AddStream to immediately create a sender, rather
than waiting until the track is seen in SDP. And the PeerConnection now
builds the list of "send streams" from the list of senders, rather than
the collection of local media streams.

Review URL: https://codereview.webrtc.org/1413713003

Cr-Commit-Position: refs/heads/master@{#10414}
2015-10-26 18:48:26 +00:00
..
androidtests
Android: Replace EGL14 with EGL10
2015-10-22 23:52:45 +00:00
java
SurfaceTextureHelper: Remove use of quitSafely() because it's API lvl 18
2015-10-24 01:17:59 +00:00
javatests
Remove references to libpeerconnection.
2015-10-08 12:41:05 +00:00
objc
talk: Use NDEBUG macro.
2015-10-22 23:15:25 +00:00
objctests
Add OWNERS for ObjC dirs.
2015-10-01 00:33:36 +00:00
test
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
androidvideocapturer.cc
passing |buffer| by reference in AndroidVideoCapturer::OnIncomingFrame
2015-10-20 18:05:02 +00:00
androidvideocapturer.h
passing |buffer| by reference in AndroidVideoCapturer::OnIncomingFrame
2015-10-20 18:05:02 +00:00
audiotrack.cc
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +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
Makes libjingle_peerconnection_android_unittest run on networkless devices.
2015-02-12 09:24:25 +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
Remove deprecated functions.
2015-07-16 20:43:27 +00:00
jsepsessiondescription.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
localaudiosource_unittest.cc
Remove use of DeviceManager from ChannelManager.
2015-09-24 07:41:59 +00:00
localaudiosource.cc
Follow-up: Remove old DelayCorrection AEC config
2015-06-16 07:53:32 +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
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
mediastreamprovider.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
mediastreamproxy.h
Update libjingle license statements at top of talk files for consistency
2015-01-20 21:36:13 +00:00
mediastreamsignaling.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
mediastreamsignaling.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
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
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
peerconnection.cc
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
peerconnection.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +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
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
peerconnectioninterface.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
peerconnectionproxy.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
portallocatorfactory.cc
Makes libjingle_peerconnection_android_unittest run on networkless devices.
2015-02-12 09:24:25 +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
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
rtpsender.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
rtpsenderinterface.h
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +00:00
rtpsenderreceiver_unittest.cc
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +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
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
statscollector.cc
Merging BaseSession code into WebRtcSession.
2015-10-14 22:02:50 +00:00
statscollector.h
Merging BaseSession code into WebRtcSession.
2015-10-14 22:02:50 +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
Remove use of DeviceManager from ChannelManager.
2015-09-24 07:41:59 +00:00
videosource.cc
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +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
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +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
Fixing some issues with the direction attribute of m-lines in offers.
2015-10-22 20:14:51 +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
Fixing some issues with the direction attribute of m-lines in offers.
2015-10-22 20:14:51 +00:00
webrtcsession.cc
Adding the ability to create an RtpSender without a track.
2015-10-26 18:48:26 +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