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fd6706a310fdf9a90e438cb9ed466f4025cf2717
platform-external-webrtc/talk/app/webrtc
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glaznev fd6706a310 Log Android HW decoder delay time statistics. 
BUG=b/26962199

Review URL: https://codereview.webrtc.org/1665373003

Cr-Commit-Position: refs/heads/master@{#11511}
2016-02-05 22:05:15 +00:00
..
androidtests
Get the adapter type information from Android OS.
2016-02-02 20:54:28 +00:00
java
Log Android HW decoder delay time statistics.
2016-02-05 22:05:15 +00:00
javatests
Remove references to libpeerconnection.
2015-10-08 12:41:05 +00:00
objc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
objctests
Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ )
2016-01-28 13:06:16 +00:00
test
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
androidvideocapturer.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
androidvideocapturer.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
datachannel.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Making WebRtcSession fire a destroyed signal.
2016-01-20 22:30:50 +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
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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
jsepsessiondescription_unittest.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
localaudiosource.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
localaudiosource.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
mediaconstraintsinterface.cc
Delete unused members from VideoOptions
2016-01-29 09:14:45 +00:00
mediaconstraintsinterface.h
Delete unused members from VideoOptions
2016-01-29 09:14:45 +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
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +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
Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface.
2016-02-04 09:24:56 +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
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnection.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnection.h
Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ )
2016-01-28 13:06:16 +00:00
peerconnectionendtoend_unittest.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnectionfactory_unittest.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnectionfactory.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnectionfactory.h
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
2016-01-15 11:06:41 +00:00
peerconnectionfactoryproxy.h
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
2016-01-15 11:06:41 +00:00
peerconnectioninterface_unittest.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnectioninterface.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
peerconnectionproxy.h
Adding AddTrack/RemoveTrack to native PeerConnection API.
2016-01-14 23:35:46 +00:00
portallocatorfactory.cc
Removing webrtc::PortAllocatorFactoryInterface.
2015-12-29 22:15:02 +00:00
portallocatorfactory.h
Removing webrtc::PortAllocatorFactoryInterface.
2015-12-29 22:15:02 +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
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
remoteaudiosource.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
remotevideocapturer.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
remotevideocapturer.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
rtpreceiver.cc
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
rtpreceiver.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
rtpreceiverinterface.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
rtpsender.cc
Deleted method AudioTrackInterface::GetRenderer.
2016-01-28 08:38:15 +00:00
rtpsender.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
rtpsenderinterface.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
rtpsenderreceiver_unittest.cc
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
statscollector.cc
https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type.
2016-01-28 09:27:20 +00:00
statscollector.h
Removed dummy "mediastreamsignaling.h"
2015-11-19 20:01:10 +00:00
statstypes.cc
https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type.
2016-01-28 09:27:20 +00:00
statstypes.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videosource.cc
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videosource.h
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videosourceinterface.h
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videosourceproxy.h
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videotrack_unittest.cc
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videotrack.cc
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videotrack.h
Delete FrameInput method and FrameInputWrapper class.
2016-02-05 09:52:20 +00:00
videotrackrenderers.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
videotrackrenderers.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtcsdp_unittest.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtcsdp.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +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
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtcsession.cc
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtcsession.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00
webrtcsessiondescriptionfactory.cc
Properly handle different transports having different SSL roles.
2016-01-08 23:36:06 +00:00
webrtcsessiondescriptionfactory.h
Move talk/media to webrtc/media
2016-02-05 07:52:35 +00:00