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philipel 8054c7ecf8 EndToEndTest.ReceivesAndRetransmitsNack now handles reordering. 
Since packets can be received out of order NACKs could be triggered before any
packet was actually dropped. This cause the test to never drop packets which in
turn caused the block of code which set the |observation_complete_| event to
never execute.

BUG=webrtc:2845

Review-Url: https://codereview.webrtc.org/2613443002
Cr-Commit-Position: refs/heads/master@{#15990}
2017-01-10 13:19:19 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Final removal of webrtc/api/android
2017-01-10 11:43:21 +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
Drop unneeded includes of base/stream.h.
2017-01-10 11:44:41 +00:00
build
Disable some filesystem tests that don't make sense on iOS.
2017-01-10 11:00:41 +00:00
call
Clean up ramp-up tests and make sure they all pass.
2017-01-09 12:19:24 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-01-10 11:37:20 +00:00
common_video
Drop unneeded includes of base/stream.h.
2017-01-10 11:44:41 +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
logging
Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
2016-12-20 13:03:58 +00:00
media
Reorder assignments in WebRtcVideoChannel2::ConfigureReceiverRtp to match definition in VideoReceiveStream::Config.
2017-01-10 13:18:15 +00:00
modules
Reland of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
2017-01-10 12:21:35 +00:00
p2p
Add disabled certificate check support to IceServer PeerConnection API.
2017-01-09 16:35:45 +00:00
pc
Reland of: Separating SCTP code from BaseChannel/MediaChannel.
2017-01-09 22:53:41 +00:00
sdk
Add a check for CameraCapturer being initialized before calling startCapture.
2017-01-10 12:30:31 +00:00
stats
RTCIceCandidateStats.transportId added and collected.
2017-01-02 17:59:31 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Disable some filesystem tests that don't make sense on iOS.
2017-01-10 11:00:41 +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
EndToEndTest.ReceivesAndRetransmitsNack now handles reordering.
2017-01-10 13:19:19 +00:00
voice_engine
Move FilePlayer and FileRecorder to Voice Engine
2017-01-10 09:12:51 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of "Remove webrtc/libjingle/{xmllite,xmpp}"
2017-01-09 07:02:01 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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.