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webrtc_m130/webrtc/pc
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zhihuang ab97e18fa9 Fix the binary size regression on Chromium Windows. 
There is a dependency chain from Chromium windows main_dll to Opus
which should never exist. We used to rely on rtc_static_library
to break this chain. So this CL replaced some rtc_source_set
with rtc_static_library.

libvpx fix (https://chromium-review.googlesource.com/c/544107/) for
ios-simulator linking issue is landed and this CL can be sumbitted once the new
Chromium is rolled into WebRTC.

BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2947273002
Cr-Commit-Position: refs/heads/master@{#18709}
2017-06-22 08:28:59 +00:00
..
test
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
audiomonitor.cc
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
audiomonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
audiotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
audiotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
BUILD.gn
Fix the binary size regression on Chromium Windows.
2017-06-22 08:28:59 +00:00
bundlefilter_unittest.cc
…
bundlefilter.cc
…
bundlefilter.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
channel_unittest.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
channel.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channel.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channelmanager_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
channelmanager.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
channelmanager.h
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
createpeerconnectionfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
currentspeakermonitor_unittest.cc
Disable flaky tests on iOS
2017-02-27 06:10:14 +00:00
currentspeakermonitor.cc
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
currentspeakermonitor.h
Reland of place basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2603203003/ )
2017-01-02 16:42:32 +00:00
datachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
datachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
DEPS
Injectable audio encoders: BuiltinAudioEncoderFactory
2017-04-06 17:03:21 +00:00
dtmfsender_unittest.cc
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +00:00
dtmfsender.cc
Change minimum DTMF event duration to be 40 milliseconds
2017-03-28 18:18:32 +00:00
dtmfsender.h
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
2017-02-02 04:27:00 +00:00
externalhmac.cc
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
externalhmac.h
Remove HAVE_SRTP define and unmaintained code.
2017-03-06 19:32:22 +00:00
iceserverparsing_unittest.cc
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
iceserverparsing.cc
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +00:00
iceserverparsing.h
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
jsepicecandidate.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
jsepsessiondescription_unittest.cc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
jsepsessiondescription.cc
Fix the fuzz test.
2017-03-23 17:32:12 +00:00
localaudiosource_unittest.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.cc
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
localaudiosource.h
Remove PC factory options param from LocalAudioSource::Create.
2017-02-11 05:26:48 +00:00
mediaconstraintsinterface_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediamonitor.cc
Delete unneeded includes of base/common.h.
2017-01-25 09:47:24 +00:00
mediamonitor.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
mediasession_unittest.cc
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
2017-05-25 16:38:55 +00:00
mediasession.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
mediasession.h
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
mediastream_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastream.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamobserver.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
mediastreamtrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
OWNERS
Make hbos and hta rtcstats* OWNERS of webrtc/pc, not webrtc/api.
2017-02-28 14:46:44 +00:00
peerconnection_integrationtest.cc
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
2017-05-25 16:38:55 +00:00
peerconnection.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnection.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectionendtoend_unittest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
peerconnectionfactory_unittest.cc
Delete deprecated PeerConnection methods, and corresponding using declarations.
2017-03-08 14:59:45 +00:00
peerconnectionfactory.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectionfactory.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
peerconnectioninterface_unittest.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
proxy_unittest.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
quicdatachannel_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatachannel.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
quicdatatransport.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
remoteaudiosource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
rtcpmuxfilter_unittest.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcpmuxfilter.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcpmuxfilter.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtcstats_integrationtest.cc
Remove temporary include of builtin_audio_encoder_factory.h.
2017-05-17 11:39:36 +00:00
rtcstatscollector_unittest.cc
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtcstatscollector.cc
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtcstatscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
rtpreceiver.cc
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
rtpreceiver.h
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
rtpsender.cc
Protect APM in webkit builds.
2017-02-03 14:37:05 +00:00
rtpsender.h
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
rtpsenderreceiver_unittest.cc
Delete MediaController class, move Call ownership to PeerConnection.
2017-05-05 09:23:02 +00:00
rtptransport_unittest.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtptransport.cc
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
rtptransport.h
Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.
2017-06-01 20:22:42 +00:00
sctputils_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
sctputils.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
srtpfilter_unittest.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
srtpfilter.cc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
srtpfilter.h
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
statscollector_unittest.cc
Fix uploading of available send bitrate statistics.
2017-06-19 15:58:15 +00:00
statscollector.cc
Fix uploading of available send bitrate statistics.
2017-06-19 15:58:15 +00:00
statscollector.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
streamcollection.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
trackmediainfomap_unittest.cc
TrackMediaInfoMap: Allow same SSRC for send and receive side.
2017-02-20 10:05:13 +00:00
trackmediainfomap.cc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
trackmediainfomap.h
Move trackmediainfomap files from api/ to pc/.
2017-02-12 03:07:31 +00:00
videocapturertracksource_unittest.cc
Switching some interfaces to use std::unique_ptr<>.
2017-02-11 04:13:37 +00:00
videocapturertracksource.cc
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
videocapturertracksource.h
Fix occasional race in VideoCapturerTrackSource seen by memcheck bot.
2017-02-25 21:37:59 +00:00
videotrack_unittest.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotrack.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.cc
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
videotracksource.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
voicechannel.h
Simple lint fixes
2016-04-26 12:28:18 +00:00
webrtcsdp_unittest.cc
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
webrtcsdp.cc
Parse the connection data in SDP (c= line).
2017-03-21 18:04:53 +00:00
webrtcsdp.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00
webrtcsession_unittest.cc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
webrtcsession.cc
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
webrtcsession.h
Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
2017-06-02 13:44:03 +00:00
webrtcsessiondescriptionfactory.cc
Fix comment about remote restart being requested in createOffer
2017-04-25 19:57:14 +00:00
webrtcsessiondescriptionfactory.h
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
2017-01-23 12:56:25 +00:00