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Johannes Kron 965e7942a3 Add sanity checks to UpdateDelayStatistics and patch unit tests. 
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.

Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
2018-09-24 23:13:02 +00:00
..
test Add sanity checks to UpdateDelayStatistics and patch unit tests. 2018-09-24 23:13:02 +00:00
audiotrack.cc Remove "using rtc::scoped_ptr" from audiotrack.cc. 2017-11-28 21:29:18 +00:00
audiotrack.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Fix no_exit_time_destructors in pc. 2018-09-11 09:32:14 +00:00
builtin_video_bitrate_allocator_factory.cc The default logic for creating video bitrate allocator. 2018-08-23 20:50:32 +00:00
builtin_video_bitrate_allocator_factory.h The default logic for creating video bitrate allocator. 2018-08-23 20:50:32 +00:00
bundlefilter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
bundlefilter.h Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" 2018-03-29 02:45:17 +00:00
bundlefilter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
channel.cc Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
channel.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
channel_unittest.cc Cleanup networkroute.h 2018-07-23 19:35:28 +00:00
channelmanager.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
channelmanager.h Use new TransportController implementation in PeerConnection. 2018-03-30 18:41:19 +00:00
channelmanager_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
createpeerconnectionfactory.cc Allows injection of network controller factory into peer connection factory. 2018-05-18 17:07:16 +00:00
datachannel.cc Use AsyncInvoker in DataChannel instead of MessageHandler 2018-08-31 22:48:20 +00:00
datachannel.h Use AsyncInvoker in DataChannel instead of MessageHandler 2018-08-31 22:48:20 +00:00
datachannel_unittest.cc Removing warning suppression flags from pc/. 2018-07-04 10:35:27 +00:00
DEPS Allows injection of network controller factory into peer connection factory. 2018-05-18 17:07:16 +00:00
dtlssrtptransport.cc Add a flag to actively reset the SRTP parameters 2018-06-12 20:32:00 +00:00
dtlssrtptransport.h Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
dtlssrtptransport_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
dtmfsender.cc Add "tones remaining" argument to DTMF ontonechange callback 2018-09-07 17:29:37 +00:00
dtmfsender.h Use AsyncInvoker in DtmfSender instead of MessageHandler 2018-08-31 22:49:26 +00:00
dtmfsender_unittest.cc Add "tones remaining" argument to DTMF ontonechange callback 2018-09-07 17:29:37 +00:00
externalhmac.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
externalhmac.h [CFI] Fix cfi-icall failures 2017-12-12 09:27:07 +00:00
iceserverparsing.cc Revert "Add SSLConfig object to IceServer." 2018-09-12 10:46:04 +00:00
iceserverparsing.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
iceserverparsing_unittest.cc Revert "Add SSLConfig object to IceServer." 2018-09-12 10:46:04 +00:00
jsepicecandidate.cc Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h 2018-08-15 15:19:07 +00:00
jsepsessiondescription.cc Add CreateSessionDescription overload which takes a cricket::SessionDescription 2018-07-25 18:03:05 +00:00
jsepsessiondescription_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
jseptransport.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
jseptransport.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
jseptransport_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
jseptransportcontroller.cc Use AsyncInvoker in JsepTransportController instead of MessageHandler 2018-08-31 22:12:45 +00:00
jseptransportcontroller.h Use AsyncInvoker in JsepTransportController instead of MessageHandler 2018-08-31 22:12:45 +00:00
jseptransportcontroller_unittest.cc Add AsyncResolverFactory interface and basic implementation. 2018-08-02 21:20:15 +00:00
localaudiosource.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
localaudiosource.h Delete deprecated CreateAudioSource method, with constraints. 2018-06-04 08:19:30 +00:00
localaudiosource_unittest.cc Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
mediaconstraintsinterface_unittest.cc Rename a few MediaConfig::Video flags for consistency. 2018-01-22 17:32:58 +00:00
mediasession.cc Use C++11 for range loop in pc/mediasession.cc (and test) 2018-09-10 20:27:34 +00:00
mediasession.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediasession_unittest.cc Make fewer copies when using StringBuilder. 2018-09-24 09:39:19 +00:00
mediastream.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastream.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
mediastreamobserver.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastreamobserver.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
mediastreamtrack.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Adding shampson (me) as an owner to pc/ & api/. 2018-07-03 20:39:17 +00:00
peerconnection.cc Make fewer copies when using StringBuilder. 2018-09-24 09:39:19 +00:00
peerconnection.h Revert "Use AsyncInvoker in PeerConnection instead of MessageHandler" 2018-09-05 21:12:18 +00:00
peerconnection_bundle_unittest.cc Add CreateIceCandidate overload which takes a cricket::Candidate 2018-07-23 23:49:44 +00:00
peerconnection_crypto_unittest.cc Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnection_datachannel_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
peerconnection_histogram_unittest.cc Modified peerconnection's "observer" slot to be nulled on close. 2018-08-01 09:44:57 +00:00
peerconnection_ice_unittest.cc Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
peerconnection_integrationtest.cc Add sanity checks to UpdateDelayStatistics and patch unit tests. 2018-09-24 23:13:02 +00:00
peerconnection_jsep_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
peerconnection_media_unittest.cc Add support for enabling simulcast in "Plan B" using MediaConstraints. 2018-08-24 09:55:59 +00:00
peerconnection_rampup_tests.cc Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnection_rtp_unittest.cc [Unified Plan] Implement FiredDirection for RtpTransceiver 2018-07-17 23:56:04 +00:00
peerconnection_signaling_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
peerconnectionendtoend_unittest.cc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
peerconnectionfactory.cc Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectionfactory.h Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectionfactory_unittest.cc Delete almost all use of MediaConstraintsInterface in the PeerConnection API 2018-08-23 07:14:37 +00:00
peerconnectioninterface_unittest.cc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
peerconnectioninternal.h Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."" 2018-07-12 21:35:47 +00:00
peerconnectionwrapper.h Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."" 2018-07-12 21:35:47 +00:00
proxy_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
remoteaudiosource.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
remoteaudiosource.h Configure media flow correctly with Unified Plan 2018-01-18 19:01:38 +00:00
rtcpmuxfilter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h Move sessiondescription.h/cc from p2p/base to pc/ 2017-12-20 00:21:52 +00:00
rtcpmuxfilter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcstats_integrationtest.cc getStats: add kind alias for mediaType 2018-09-03 11:49:30 +00:00
rtcstatscollector.cc getStats: add kind alias for mediaType 2018-09-03 11:49:30 +00:00
rtcstatscollector.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
rtcstatscollector_unittest.cc getStats: add kind alias for mediaType 2018-09-03 11:49:30 +00:00
rtcstatstraversal.cc RTCStatsCollector::GetStatsReport() with optional selector argument. 2018-03-19 15:32:16 +00:00
rtcstatstraversal.h Stats traversal algorithm added. 2018-03-12 10:54:09 +00:00
rtcstatstraversal_unittest.cc Stats traversal algorithm added. 2018-03-12 10:54:09 +00:00
rtpmediautils.cc Implement legacy offer_to_receive options for Unified Plan 2018-01-27 02:20:29 +00:00
rtpmediautils.h Reland "Reland "Remove our stream << overloads from non-test build targets."" 2018-05-03 10:41:41 +00:00
rtpmediautils_unittest.cc Implement legacy offer_to_receive options for Unified Plan 2018-01-27 02:20:29 +00:00
rtpparametersconversion.cc Add PeerConnection GetRtpSender/ReceiverCapabilities 2018-06-28 12:40:07 +00:00
rtpparametersconversion.h Add PeerConnection GetRtpSender/ReceiverCapabilities 2018-06-28 12:40:07 +00:00
rtpparametersconversion_unittest.cc Add PeerConnection GetRtpSender/ReceiverCapabilities 2018-06-28 12:40:07 +00:00
rtpreceiver.cc Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. 2018-09-11 20:10:44 +00:00
rtpreceiver.h Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. 2018-09-11 20:10:44 +00:00
rtpsender.cc Reland "Add initial support for RtpEncodingParameters max_framerate." 2018-09-13 10:06:33 +00:00
rtpsender.h Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel. 2018-09-11 20:10:44 +00:00
rtpsenderreceiver_unittest.cc Reland "Add initial support for RtpEncodingParameters max_framerate." 2018-09-13 10:06:33 +00:00
rtptransceiver.cc [Unified Plan] Implement FiredDirection for RtpTransceiver 2018-07-17 23:56:04 +00:00
rtptransceiver.h [Unified Plan] Implement FiredDirection for RtpTransceiver 2018-07-17 23:56:04 +00:00
rtptransport.cc Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
rtptransport.h Delete struct webrtc::PacketTime. 2018-08-07 10:07:15 +00:00
rtptransport_unittest.cc Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
rtptransportinternal.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
rtptransportinternaladapter.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
rtptransporttestutil.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
sctputils.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
sctputils.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
sctputils_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
sdputils.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
sdputils.h Move sessiondescription.h/cc from p2p/base to pc/ 2017-12-20 00:21:52 +00:00
sessiondescription.cc Use the SDP ContentInfo helpers to avoid downcasting 2017-12-22 00:17:53 +00:00
sessiondescription.h Add MSID signaling compatibility for Unified Plan endpoints 2018-02-01 21:17:41 +00:00
srtpfilter.cc Move base64.h to the proper location. 2018-07-23 15:40:36 +00:00
srtpfilter.h Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
srtpfilter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
srtpsession.cc Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
srtpsession.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
srtpsession_unittest.cc Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."" 2018-07-12 21:35:47 +00:00
srtptestutil.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
srtptransport.cc Move base64.h to the proper location. 2018-07-23 15:40:36 +00:00
srtptransport.h Remove MetricsObserverInterface. 2018-07-19 23:00:20 +00:00
srtptransport_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
statscollector.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
statscollector.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
statscollector_unittest.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
streamcollection.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
trackmediainfomap.cc Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
trackmediainfomap.h Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
trackmediainfomap_unittest.cc Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
transportstats.cc Create the JsepTransportController and JsepTransport2. 2018-02-23 00:13:45 +00:00
transportstats.h Create the JsepTransportController and JsepTransport2. 2018-02-23 00:13:45 +00:00
videocapturertracksource.cc Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
videocapturertracksource.h Move sigslot to proper third_party directory 2018-07-25 14:53:33 +00:00
videocapturertracksource_unittest.cc Reimplement rtc::ToString and rtc::FromString without streams. 2018-08-16 16:14:01 +00:00
videotrack.cc Enable cpplint in pc/ 2017-10-30 18:08:29 +00:00
videotrack.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
videotrack_unittest.cc Refactor VideoTrackSource, without raw pointer injection. 2018-05-23 15:42:10 +00:00
videotracksource.cc Delete deprecated VideoTrackSource constructor. 2018-06-11 19:15:10 +00:00
videotracksource.h Replace rtc::Optional with absl::optional in pc 2018-06-19 20:55:07 +00:00
webrtcsdp.cc Make HasAttribute handle partial matching of attribute names. 2018-09-10 08:43:52 +00:00
webrtcsdp.h Add MSID signaling compatibility for Unified Plan endpoints 2018-02-01 21:17:41 +00:00
webrtcsdp_unittest.cc Make HasAttribute handle partial matching of attribute names. 2018-09-10 08:43:52 +00:00
webrtcsessiondescriptionfactory.cc Reland "Delete leftover includes and declarations for MediaConstraintsInterface" 2018-09-03 09:00:01 +00:00
webrtcsessiondescriptionfactory.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00