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Taylor Brandstetter 80cfb527c6 RTC_CHECK'ing content type before static_casting descriptions. 
This will cause the application to be aborted before it encounters
something worse like a heap overflow, in case any bug in this code
exists or is introduced in the future.

TBR=zhihuang@webrtc.org

Bug: chromium:773620
Change-Id: Idd4e31aa63a3f673eefd3e8cb2ae3f4a5092ca4e
Reviewed-on: https://webrtc-review.googlesource.com/9040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20293}
2017-10-13 16:36:28 +00:00
..
test
Reland "Added PeerConnectionObserver::OnRemoveTrack."
2017-10-10 17:06:00 +00:00
audiomonitor.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
audiomonitor.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
audiotrack.cc
Add explicit includes of refcountedobject.h where it is used.
2017-10-06 13:00:14 +00:00
audiotrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
BUILD.gn
PeerConnection end-to-end test with a non-builtin codec
2017-10-11 11:48:54 +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
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
channel.cc
Change the flag when RtpTransport objects send packet.
2017-10-05 04:41:55 +00:00
channel.h
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
2017-10-06 18:50:24 +00:00
channelmanager_unittest.cc
Move the TransportController from p2p/base to pc/.
2017-09-29 18:20:07 +00:00
channelmanager.cc
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
2017-10-13 05:11:27 +00:00
channelmanager.h
Delete unneeded includes of fileutils.h
2017-09-29 12:39:09 +00:00
createpeerconnectionfactory.cc
Remove unnecessary video factory references in PeerConnectionFactory
2017-09-27 14:41:46 +00:00
currentspeakermonitor_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
currentspeakermonitor.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
currentspeakermonitor.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
datachannel_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
datachannel.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
datachannel.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
dtmfsender_unittest.cc
Clean up dependencies of peerconnection_unittest.
2017-09-15 12:51:00 +00:00
dtmfsender.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
dtmfsender.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
externalhmac.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
externalhmac.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
iceserverparsing_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
iceserverparsing.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
jsepsessiondescription.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
localaudiosource_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediamonitor.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediamonitor.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediasession_unittest.cc
Removing logging in unit test that was committed accidentally.
2017-10-04 02:32:06 +00:00
mediasession.cc
RTC_CHECK'ing content type before static_casting descriptions.
2017-10-13 16:36:28 +00:00
mediasession.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastream.cc
Add explicit includes of refcountedobject.h where it is used.
2017-10-06 13:00:14 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
peerconnection_crypto_unittest.cc
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
2017-10-06 18:50:24 +00:00
peerconnection_integrationtest.cc
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
peerconnection_rtp_unittest.cc
Reland "Added PeerConnectionObserver::OnRemoveTrack."
2017-10-10 17:06:00 +00:00
peerconnection.cc
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
peerconnection.h
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
peerconnectionendtoend_unittest.cc
PeerConnection end-to-end test with a non-builtin codec
2017-10-11 11:48:54 +00:00
peerconnectionfactory_unittest.cc
Clean up dependencies of peerconnection_unittest.
2017-09-15 12:51:00 +00:00
peerconnectionfactory.cc
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
2017-10-11 07:45:29 +00:00
peerconnectionfactory.h
Remove dead version of StartRtcEventLog
2017-10-06 15:18:24 +00:00
peerconnectioninterface_unittest.cc
Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed
2017-10-11 17:34:17 +00:00
peerconnectionwrapper.cc
Add sdputils.h with useful functions for working with session descriptions
2017-10-06 17:42:24 +00:00
peerconnectionwrapper.h
Add sdputils.h with useful functions for working with session descriptions
2017-10-06 17:42:24 +00:00
proxy_unittest.cc
Clean up dependencies of peerconnection_unittest.
2017-09-15 12:51:00 +00:00
quicdatachannel_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quicdatachannel.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quicdatachannel.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quicdatatransport_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quicdatatransport.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
quicdatatransport.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
remoteaudiosource.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
remoteaudiosource.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcpmuxfilter.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcstats_integrationtest.cc
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
2017-10-02 10:47:00 +00:00
rtcstatscollector_unittest.cc
Move clients of WebRtcSession to use PeerConnection
2017-10-02 17:16:09 +00:00
rtcstatscollector.cc
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
rtcstatscollector.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpreceiver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpreceiver.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpsender.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpsender.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtpsenderreceiver_unittest.cc
Move the TransportController from p2p/base to pc/.
2017-09-29 18:20:07 +00:00
rtptransport_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtptransport.cc
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
rtptransport.h
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
rtptransportinternal.h
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sctputils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sdputils.cc
Add sdputils.h with useful functions for working with session descriptions
2017-10-06 17:42:24 +00:00
sdputils.h
Add sdputils.h with useful functions for working with session descriptions
2017-10-06 17:42:24 +00:00
srtpfilter_unittest.cc
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
srtpfilter.cc
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
2017-10-13 05:11:27 +00:00
srtpfilter.h
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
2017-10-13 05:11:27 +00:00
srtpsession_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtpsession.cc
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
2017-10-13 05:11:27 +00:00
srtpsession.h
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
2017-10-13 05:11:27 +00:00
srtptestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtptransport_unittest.cc
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
srtptransport.cc
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
srtptransport.h
Reland: Completed the functionalities of SrtpTransport.
2017-09-26 18:12:45 +00:00
statscollector_unittest.cc
Move clients of WebRtcSession to use PeerConnection
2017-10-02 17:16:09 +00:00
statscollector.cc
Move clients of WebRtcSession to use PeerConnection
2017-10-02 17:16:09 +00:00
statscollector.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
streamcollection.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
trackmediainfomap_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
trackmediainfomap.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
trackmediainfomap.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
transportcontroller_unittest.cc
Move the TransportController from p2p/base to pc/.
2017-09-29 18:20:07 +00:00
transportcontroller.cc
Move the TransportController from p2p/base to pc/.
2017-09-29 18:20:07 +00:00
transportcontroller.h
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
videocapturertracksource_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
videocapturertracksource.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Add explicit includes of refcountedobject.h where it is used.
2017-10-06 13:00:14 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
voicechannel.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
webrtcsdp_unittest.cc
Serialize "a=x-google-flag:conference".
2017-09-16 08:24:29 +00:00
webrtcsdp.cc
Reland "Clean up libjingle API dependencies."
2017-10-05 13:51:21 +00:00
webrtcsdp.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
webrtcsession_unittest.cc
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
2017-10-06 18:50:24 +00:00
webrtcsession.cc
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
2017-10-06 18:50:24 +00:00
webrtcsession.h
Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests
2017-10-06 18:50:24 +00:00
webrtcsessiondescriptionfactory.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
webrtcsessiondescriptionfactory.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00