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Harald Alvestrand 76d295231a Don't crash when sender info has been discarded by lower layers. 
This happens when pc.close() is called.
As a stopgap measure, we return zeroes instead, leading to stats
being omitted.

Bug: chromium:807174
Change-Id: I36f342adcd038822afb75d8593de808591eb9c4b
Reviewed-on: https://webrtc-review.googlesource.com/46161
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21813}
2018-01-30 18:04:06 +00:00
..
test
Rewrite StatsCollector tests to use a fake PeerConnection
2018-01-26 19:40:02 +00:00
audiomonitor.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +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
Break up rtc_event_log_api to solve circular dependencies.
2018-01-30 17:54:06 +00:00
bundlefilter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
bundlefilter.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
bundlefilter.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
channel_unittest.cc
Delete MediaMonitor.
2018-01-17 10:33:55 +00:00
channel.cc
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
channel.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
channelmanager_unittest.cc
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
channelmanager.cc
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
channelmanager.h
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
createpeerconnectionfactory.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +00:00
currentspeakermonitor_unittest.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
currentspeakermonitor.cc
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
currentspeakermonitor.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
datachannel_unittest.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
datachannel.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
datachannel.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
dtlssrtptransport_unittest.cc
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
dtlssrtptransport.cc
Use the DtlsSrtpTransport in BaseChannel.
2017-11-29 19:38:39 +00:00
dtlssrtptransport.h
Make the DtlsSrtpTransport cache the RtpAbsSendTimeHeaderExtension.
2017-11-30 02:17:09 +00:00
dtmfsender_unittest.cc
Clean up dependencies of peerconnection_unittest.
2017-09-15 12:51:00 +00:00
dtmfsender.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
dtmfsender.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
externalhmac.cc
Remove unnecessary "memset".
2017-12-22 00:22:23 +00:00
externalhmac.h
[CFI] Fix cfi-icall failures
2017-12-12 09:27:07 +00:00
iceserverparsing_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
iceserverparsing.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
iceserverparsing.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepicecandidate.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepsessiondescription_unittest.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
jsepsessiondescription.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
jseptransport_unittest.cc
Reland "Move JsepTransport from p2p/base to pc/."
2017-12-19 20:50:41 +00:00
jseptransport.cc
Reland "Move JsepTransport from p2p/base to pc/."
2017-12-19 20:50:41 +00:00
jseptransport.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
localaudiosource_unittest.cc
Optional: Use nullopt and implicit construction in /pc
2017-11-21 17:53:37 +00:00
localaudiosource.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
localaudiosource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediaconstraintsinterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
mediasession_unittest.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
mediasession.cc
Parameterize PeerConnection media tests for Unified Plan
2018-01-25 01:34:46 +00:00
mediasession.h
Parameterize PeerConnection media tests for Unified Plan
2018-01-25 01:34:46 +00:00
mediastream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastream.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
mediastream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamobserver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamobserver.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamtrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Make hbos@webrtc.org OWNER of peerconnection*.
2017-11-13 12:27:29 +00:00
peerconnection_bundle_unittest.cc
Parameterize PeerConnection BUNDLE tests for Unified Plan
2018-01-19 20:32:23 +00:00
peerconnection_crypto_unittest.cc
Parameterize PeerConnection crypto tests for Unified Plan
2018-01-19 23:28:53 +00:00
peerconnection_datachannel_unittest.cc
Parameterize PeerConnection data channel tests for Unified Plan
2018-01-20 00:50:53 +00:00
peerconnection_ice_unittest.cc
Revert "Parameterize PeerConnection signaling tests for Unified Plan"
2018-01-26 22:44:30 +00:00
peerconnection_integrationtest.cc
Implemented the GetRemoteAudioSSLCertificate method.
2018-01-27 23:48:36 +00:00
peerconnection_jsep_unittest.cc
Add PeerConnection interop integration tests
2018-01-26 23:24:49 +00:00
peerconnection_media_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
peerconnection_rtp_unittest.cc
Test creating two senders with the same track.
2018-01-17 12:49:56 +00:00
peerconnection_signaling_unittest.cc
Revert "Parameterize PeerConnection signaling tests for Unified Plan"
2018-01-26 22:44:30 +00:00
peerconnection.cc
Implemented the GetRemoteAudioSSLCertificate method.
2018-01-27 23:48:36 +00:00
peerconnection.h
Implemented the GetRemoteAudioSSLCertificate method.
2018-01-27 23:48:36 +00:00
peerconnectionendtoend_unittest.cc
Parameterize PeerConnection end to end tests for Unified Plan
2018-01-25 04:03:18 +00:00
peerconnectionfactory_unittest.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
peerconnectionfactory.cc
Revert "Remove nogncheck and add proper dependencies."
2018-01-18 19:15:49 +00:00
peerconnectionfactory.h
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
peerconnectioninterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
peerconnectioninternal.h
Add FakePeerConnectionBase
2018-01-25 01:04:06 +00:00
peerconnectionwrapper.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
peerconnectionwrapper.h
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
proxy_unittest.cc
Clean up dependencies of peerconnection_unittest.
2017-09-15 12:51:00 +00:00
remoteaudiosource.cc
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
remoteaudiosource.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtcpmuxfilter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtcpmuxfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
rtcstats_integrationtest.cc
Generate signed packets_lost in WebRTC-stats
2017-12-13 12:25:42 +00:00
rtcstatscollector_unittest.cc
Don't crash when sender info has been discarded by lower layers.
2018-01-30 18:04:06 +00:00
rtcstatscollector.cc
Don't crash when sender info has been discarded by lower layers.
2018-01-30 18:04:06 +00:00
rtcstatscollector.h
Move internal PeerConnection methods to PeerConnectionInternal
2018-01-24 17:24:29 +00:00
rtpmediautils_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.h
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpreceiver.cc
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtpreceiver.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtpsender.cc
Change return type of RtpSenderInterface::SetParameters from bool to RTCError
2018-01-25 19:06:04 +00:00
rtpsender.h
Change return type of RtpSenderInterface::SetParameters from bool to RTCError
2018-01-25 19:06:04 +00:00
rtpsenderreceiver_unittest.cc
Change return type of RtpSenderInterface::SetParameters from bool to RTCError
2018-01-25 19:06:04 +00:00
rtptransceiver.cc
Change RtpSenders to interact with the media channel directly
2018-01-13 01:44:04 +00:00
rtptransceiver.h
Change RtpReceivers to interact with the media channel directly
2018-01-11 00:16:44 +00:00
rtptransport_unittest.cc
Fixing some clang-tidy findings.
2017-11-21 16:43:07 +00:00
rtptransport.cc
Use the DtlsSrtpTransport in BaseChannel.
2017-11-29 19:38:39 +00:00
rtptransport.h
Use the DtlsSrtpTransport in BaseChannel.
2017-11-29 19:38:39 +00:00
rtptransportinternal.h
Use the DtlsSrtpTransport in BaseChannel.
2017-11-29 19:38:39 +00:00
rtptransportinternaladapter.h
Created the DtlsSrtpTransport.
2017-11-20 23:18:22 +00:00
rtptransporttestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sctputils_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sctputils.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
sctputils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sdputils.cc
Replace string type with SdpType enum
2017-12-07 19:08:47 +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
Fix shadowed variable in ContentInfo
2018-01-05 21:12:09 +00:00
srtpfilter_unittest.cc
Fix ortc_api circular deps.
2017-11-15 13:31:51 +00:00
srtpfilter.cc
Optional: Use nullopt and implicit construction in /pc
2017-11-21 17:53:37 +00:00
srtpfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
srtpsession_unittest.cc
Remove the SetEncryptedHeaderExtensionIds methods.
2017-11-11 01:14:35 +00:00
srtpsession.cc
Remove the SetEncryptedHeaderExtensionIds methods.
2017-11-11 01:14:35 +00:00
srtpsession.h
Remove the SetEncryptedHeaderExtensionIds methods.
2017-11-11 01:14:35 +00:00
srtptestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtptransport_unittest.cc
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
srtptransport.cc
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
srtptransport.h
Prepare to make BaseChannel depend on RtpTransportInternal only.
2017-12-04 22:27:39 +00:00
statscollector_unittest.cc
Rewrite StatsCollector tests to use a fake PeerConnection
2018-01-26 19:40:02 +00:00
statscollector.cc
Move internal PeerConnection methods to PeerConnectionInternal
2018-01-24 17:24:29 +00:00
statscollector.h
Move internal PeerConnection methods to PeerConnectionInternal
2018-01-24 17:24:29 +00:00
streamcollection.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
trackmediainfomap_unittest.cc
Reland "Move stats ID generation from SSRC to local ID"
2018-01-11 18:04:22 +00:00
trackmediainfomap.cc
Reland "Move stats ID generation from SSRC to local ID"
2018-01-11 18:04:22 +00:00
trackmediainfomap.h
Reland "Move stats ID generation from SSRC to local ID"
2018-01-11 18:04:22 +00:00
transportcontroller_unittest.cc
Refactor stream stats generation
2018-01-14 09:35:07 +00:00
transportcontroller.cc
Implemented the GetRemoteAudioSSLCertificate method.
2018-01-27 23:48:36 +00:00
transportcontroller.h
Implemented the GetRemoteAudioSSLCertificate method.
2018-01-27 23:48:36 +00:00
videocapturertracksource_unittest.cc
Optional: Use nullopt and implicit construction in /pc
2017-11-21 17:53:37 +00:00
videocapturertracksource.cc
Optional: Use nullopt and implicit construction in /pc
2017-11-21 17:53:37 +00:00
videocapturertracksource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videotrack_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
videotracksource.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videotracksource.h
Move videosinkinterface.h to common_video to solve a circular dep.
2018-01-04 13:19:49 +00:00
webrtcsdp_unittest.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
webrtcsdp.cc
Use the SDP ContentInfo helpers to avoid downcasting
2017-12-22 00:17:53 +00:00
webrtcsdp.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
webrtcsessiondescriptionfactory.cc
Move internal PeerConnection methods to PeerConnectionInternal
2018-01-24 17:24:29 +00:00
webrtcsessiondescriptionfactory.h
Move internal PeerConnection methods to PeerConnectionInternal
2018-01-24 17:24:29 +00:00