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zhihuang 1a619cdb5d Create the MockIceTransport 
The src/remoting/protocol/channel_socket_adapter_unittest.cc will use this class
instead of creating its own MockTransportChannel which subclasses the TransportChannel.
This will make it easier to switch the base class of P2PTransportChannel from
TransportChannel to IceTransportInternal.

BUG=webrtc:6951

Review-Url: https://codereview.webrtc.org/2610663003
Cr-Commit-Position: refs/heads/master@{#15922}
2017-01-05 20:21:52 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
2017-01-05 04:28:21 +00:00
audio
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.
2017-01-05 14:03:24 +00:00
base
GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.
2017-01-05 14:03:24 +00:00
build
Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ )
2017-01-05 12:24:37 +00:00
call
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
2017-01-05 15:09:50 +00:00
common_audio
Fixing init time error in smoothing filter.
2016-12-28 10:57:50 +00:00
common_video
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
examples
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
libjingle
Revert of Remove webrtc/libjingle/{xmllite,xmpp} (patchset #1 id:1 of https://codereview.webrtc.org/2617443003/ )
2017-01-05 06:29:39 +00:00
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
2017-01-05 04:28:21 +00:00
modules
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
2017-01-05 15:09:50 +00:00
p2p
Create the MockIceTransport
2017-01-05 20:21:52 +00:00
pc
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
2017-01-05 04:28:21 +00:00
sdk
Revert of Fixing package-boundary violation with srjar_deps (patchset #5 id:80001 of https://codereview.webrtc.org/2610823002/ )
2017-01-05 12:24:37 +00:00
stats
RTCIceCandidateStats.transportId added and collected.
2017-01-02 17:59:31 +00:00
system_wrappers
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
test
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
tools
Add method needed to extract frame capture and arrival timestamps from rtc event logs.
2016-12-20 16:51:52 +00:00
video
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
2017-01-05 15:09:50 +00:00
voice_engine
Drop unneeded include of media_file.h.
2016-12-21 11:05:44 +00:00
.gitignore
…
BUILD.gn
Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )
2017-01-05 15:09:50 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove OverUseDetectorOptions from OveruseDetector since it isn't used.
2016-12-27 18:43:01 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.