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webrtc/api
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Jiawei Ou 55718120e6 Adding rtcp report interval into RTCConfiguration. 
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.

Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.

Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
2018-11-12 20:00:00 +00:00
..
audio AEC3: Decrease latency until the delay has been detected 2018-10-31 07:29:48 +00:00
audio_codecs [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
crypto [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
ortc Remove most of api/ortc/. 2018-11-12 11:24:07 +00:00
stats Export symbols needed by the Chromium component build (part 3). 2018-10-16 12:57:04 +00:00
test Implement data channels over media transport. 2018-11-09 00:40:32 +00:00
transport Removes deprecated GetSentPacket from PacketResult. 2018-11-06 19:03:13 +00:00
units [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
video Pass HdrMetadata between VideoFrame and EncodedImage for VP9 2018-11-09 13:33:37 +00:00
video_codecs Add ability to specify if rate controller of video encoder is trusted. 2018-11-08 16:41:12 +00:00
array_view.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
array_view_unittest.cc ArrayView, adding ctor for fixed-size views of const(expr) std::array. 2018-05-15 13:49:02 +00:00
asyncresolverfactory.h Support domain name ICE candidates 2018-08-24 04:54:43 +00:00
audio_options.cc Remove AECM comfort noise setting from API 2018-10-16 09:42:16 +00:00
audio_options.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
bitrate_constraints.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
BUILD.gn Remove most of api/ortc/. 2018-11-12 11:24:07 +00:00
candidate.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
candidate.h Export symbols needed by the Chromium component build (part 2). 2018-10-15 19:52:31 +00:00
cryptoparams.h Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" 2018-10-11 23:09:07 +00:00
datachannelinterface.cc Enabling clang::find_bad_constructs for libjingle_peerconnection_api. 2018-07-19 09:17:10 +00:00
datachannelinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
DEPS Callback changes to media transport interface: 2018-11-08 17:47:09 +00:00
dtmfsenderinterface.h Add "tones remaining" argument to DTMF ontonechange callback 2018-09-07 17:29:37 +00:00
fec_controller.h Revert "Revert "Enables PeerConnectionFactory using external fec controller"" 2018-02-20 12:41:55 +00:00
jsep.cc Move SdpType from/to string definition close to declaration. 2018-10-12 09:59:40 +00:00
jsep.h Export symbols needed by the Chromium component build (part 6). 2018-10-23 06:48:51 +00:00
jsepicecandidate.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
jsepicecandidate.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
jsepsessiondescription.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
media_transport_interface.cc Callback changes to media transport interface: 2018-11-08 17:47:09 +00:00
media_transport_interface.h Callback changes to media transport interface: 2018-11-08 17:47:09 +00:00
mediaconstraintsinterface.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
mediaconstraintsinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
mediastreaminterface.cc Remove deprecated AudioProcessing::GetStatistics function 2018-11-01 11:21:15 +00:00
mediastreaminterface.h Remove deprecated AudioProcessing::GetStatistics function 2018-11-01 11:21:15 +00:00
mediastreamproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastreamtrackproxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediatypes.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediatypes.h Convert video quality test from a TEST_F to a TEST fixture. 2018-06-21 15:49:43 +00:00
notifier.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Add owners for media_transport_interface 2018-11-08 17:45:39 +00:00
peerconnectionfactoryproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectioninterface.cc Compute RTCConnectionState and RTCIceConnectionState. 2018-10-22 11:33:17 +00:00
peerconnectioninterface.h Adding rtcp report interval into RTCConfiguration. 2018-11-12 20:00:00 +00:00
peerconnectionproxy.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
proxy.cc Add default constructor for rtc::Event 2018-11-07 08:57:50 +00:00
proxy.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
refcountedbase.h New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
rtcerror.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror.h Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtcerror_unittest.cc Remove all remaining non-test uses of std::stringstream. 2018-09-13 08:52:05 +00:00
rtceventlogoutput.h Move RtcEventLogOutput to api/ 2017-10-06 13:58:14 +00:00
rtp_headers.cc Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtp_headers.h Add RTP header extension for HDR metadata 2018-11-09 11:10:12 +00:00
rtpparameters.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtpparameters.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpreceiverinterface.cc Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpreceiverinterface.h Injects FrameEncryptorInterface into RtpSender. 2018-08-30 00:33:54 +00:00
rtpsenderinterface.cc Add support for send_encodings parameters in addTransceiver 2018-10-01 22:56:30 +00:00
rtpsenderinterface.h Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
rtptransceiverinterface.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
rtptransceiverinterface.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
setremotedescriptionobserverinterface.h Reland "SetRemoteDescriptionObserverInterface added." 2017-11-23 19:59:48 +00:00
statstypes.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
statstypes.h Revert "Add framesRendered to StatsReport" 2018-07-27 14:53:07 +00:00
turncustomizer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
umametrics.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
videosourceproxy.h Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00