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webrtc_m130/webrtc/api
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hbos 6769c49418 RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added. 
Recently added to the spec:
RTCRTPStreamStats.qpSum - https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
RTCInboundRTPStreamStats.framesDecoded - https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdecoded
RTCOutboundRTPStreamStats.framesEncoded - https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framesencoded

These are added and collected. However, the qpSum is only collected in
the outbound case. It should be collected in the inbound case before
closing crbug.com/657855

BUG=chromium:657854, chromium:657855, chromium:657856

Review-Url: https://codereview.webrtc.org/2588373005
Cr-Commit-Position: refs/heads/master@{#15872}
2017-01-02 16:35:13 +00:00
..
android/jni
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
audio
Changed the interface AudioMixer::RemoveSource to have a void return type.
2016-11-18 10:03:08 +00:00
call
Move FlexfecReceiveStream from api/call/ to call/.
2016-12-19 09:13:46 +00:00
stats
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
2017-01-02 16:35:13 +00:00
test
RTCStatsCollectorTest: Remove ExpectReportContainsDataChannel.
2016-12-21 09:57:46 +00:00
video
New gn target video_frame_api.
2016-12-02 12:01:21 +00:00
audiotrack.cc
Propagate MediaStreamSource state to video tracks the same way as audio.
2016-03-23 07:34:01 +00:00
audiotrack.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
BUILD.gn
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
2016-12-22 18:53:38 +00:00
datachannel_unittest.cc
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannel.cc
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannel.h
Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
2016-11-14 09:41:56 +00:00
datachannelinterface.h
DataChannelInterface: Remove default implementation of methods.
2016-11-24 09:18:00 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
dtmfsender_unittest.cc
Fixing flaky DtmfSenderTest by using fake clock.
2016-10-28 20:53:17 +00:00
dtmfsender.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
dtmfsender.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
dtmfsenderinterface.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
fakemediacontroller.h
Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection.
2016-03-01 20:42:08 +00:00
fakemetricsobserver.cc
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
fakemetricsobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
jsep.h
Replace basictypes.h with stddef.h for size_t.
2017-01-02 14:44:41 +00:00
jsepicecandidate.cc
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepicecandidate.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
jsepsessiondescription_unittest.cc
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
jsepsessiondescription.cc
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
jsepsessiondescription.h
Remove cricket::VideoCodec with, height and framerate properties
2016-10-24 08:21:24 +00:00
localaudiosource_unittest.cc
This CL provides interfaces that do not use constraints for
2016-03-04 10:51:44 +00:00
localaudiosource.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
localaudiosource.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
mediaconstraintsinterface_unittest.cc
Embed a cricket::MediaConfig in RTCConfiguration.
2016-04-12 06:25:34 +00:00
mediaconstraintsinterface.cc
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediaconstraintsinterface.h
Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-11 10:18:00 +00:00
mediacontroller.cc
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
mediacontroller.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +00:00
mediastream_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
mediastream.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastream.h
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
mediastreaminterface.h
Replace basictypes.h with stddef.h for size_t.
2017-01-02 14:44:41 +00:00
mediastreamobserver.cc
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamobserver.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
mediastreamproxy.h
Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP.
2016-04-15 10:49:13 +00:00
mediastreamtrack.h
Removed MediaStreamTrackInterface::set_state
2016-03-24 10:16:23 +00:00
mediastreamtrackproxy.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
notifier.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
OWNERS
Move androidvideotracksource from api under api/android/jni.
2016-11-22 09:26:22 +00:00
peerconnection_unittest.cc
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
peerconnection.cc
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
peerconnection.h
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
peerconnectionendtoend_unittest.cc
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define.
2016-12-22 18:53:38 +00:00
peerconnectionfactory_unittest.cc
Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 13:17:31 +00:00
peerconnectionfactory.cc
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactory.h
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
peerconnectionfactoryproxy.h
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
peerconnectioninterface_unittest.cc
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
peerconnectioninterface.h
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
peerconnectionproxy.h
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
2016-12-24 09:43:32 +00:00
proxy_unittest.cc
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
proxy.h
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
quicdatachannel_unittest.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.cc
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatachannel.h
Refactoring on QUIC related classes.
2016-07-13 21:13:56 +00:00
quicdatatransport_unittest.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.cc
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
quicdatatransport.h
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
remoteaudiosource.cc
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
remoteaudiosource.h
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
rtcstats_integrationtest.cc
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
2017-01-02 16:35:13 +00:00
rtcstatscollector_unittest.cc
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
2017-01-02 16:35:13 +00:00
rtcstatscollector.cc
RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
2017-01-02 16:35:13 +00:00
rtcstatscollector.h
Removed RTCStatsCollector::ProducePartialResultsOnWorkerThread.
2017-01-02 12:28:51 +00:00
rtpparameters.h
Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters.
2016-06-22 07:46:19 +00:00
rtpreceiver.cc
Create the Java Wrapper of RtpReceiverObserverInterface.
2016-12-07 18:36:49 +00:00
rtpreceiver.h
Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
2017-01-02 15:32:25 +00:00
rtpreceiverinterface.h
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
2016-06-27 23:30:45 +00:00
rtpsender.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
rtpsender.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
rtpsenderinterface.h
Separating internal and external methods of RtpSender/RtpReceiver.
2016-06-06 21:27:43 +00:00
rtpsenderreceiver_unittest.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
sctputils_unittest.cc
Reinstated sctputils_unittest.cc
2016-11-08 10:05:36 +00:00
sctputils.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
sctputils.h
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
statscollector_unittest.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
statscollector.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
statscollector.h
Return false if PeerConnection::GetStats() is called on invalid tracks
2016-11-04 18:38:19 +00:00
statstypes.cc
remove googViewLimitedResolution stat
2016-12-12 12:46:27 +00:00
statstypes.h
Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2581663002/ )
2016-12-15 11:54:52 +00:00
streamcollection.h
Fix license headers in webrtc/api.
2016-02-10 15:54:53 +00:00
umametrics.h
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
videocapturertracksource_unittest.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videocapturertracksource.cc
Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ )
2016-10-24 07:22:39 +00:00
videocapturertracksource.h
Remove stop method from VideoTrackSourceInterface.
2016-09-06 09:25:11 +00:00
videosourceproxy.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotrack_unittest.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotrack.cc
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
videotrack.h
Add support for content hints to VideoTrack.
2016-12-16 23:39:11 +00:00
videotracksource.cc
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
videotracksource.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
webrtcsdp_unittest.cc
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.cc
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
webrtcsdp.h
Reland the CL to remove candidates when doing continual gathering
2016-03-14 18:59:34 +00:00
webrtcsession_unittest.cc
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
webrtcsession.cc
Implement current/pending session description methods.
2016-12-21 01:56:17 +00:00
webrtcsession.h
Implement current/pending session description methods.
2016-12-21 01:56:17 +00:00
webrtcsessiondescriptionfactory.cc
Implement the "needs-ice-restart" logic for SetConfiguration.
2016-12-10 21:15:39 +00:00
webrtcsessiondescriptionfactory.h
Remove DtlsIdentityStoreInterface, it is no longer used.
2016-07-11 11:10:14 +00:00