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b10f32f9b24a7c0e3971b150d2d835c41e685c95
platform-external-webrtc/webrtc
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deadbeef b10f32f9b2 Adding more comments to every header file in api/ subdirectory. 
Many of these interfaces are not intuitive, or are the way they are for
complex historical reasons, so it would be nice to document these things
for future developers.

Also, many nonstandard things (such as RTCConfiguration options) were
not documented at all before this CL.

BUG=webrtc:7131
TBR=pthatcher@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2680273002
Cr-Commit-Position: refs/heads/master@{#16485}
2017-02-08 09:38:21 +00:00
..
api
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
audio
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
base
Replace all use of the VERIFY macro.
2017-02-07 15:18:43 +00:00
call
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
2017-02-07 09:18:43 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Delete VideoReceiveStream::Config::pre_render_callback.
2017-02-07 17:37:41 +00:00
examples
Replace all use of the VERIFY macro.
2017-02-07 15:18:43 +00:00
logging
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +00:00
media
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
modules
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
p2p
Replace all use of the VERIFY macro.
2017-02-07 15:18:43 +00:00
pc
Fix the Chrome crash caused by RtcEventLog
2017-02-07 23:45:16 +00:00
sdk
Add chromium-junit4 tag to instrumentation test AndroidManifests.
2017-02-07 12:12:30 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
tools
Remove all occurrences of "using std::string".
2017-02-02 16:10:00 +00:00
video
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
voice_engine
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
.gitignore
…
BUILD.gn
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.