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webrtc_m130/webrtc
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Steve Anton b3329179a7 Rename RsidResolutionObserver to SsrcBindingObserver. 
This rename prepares SsrcBindingObserver to be used for observing all
the ways a sink can be bound to SSRCs (e.g., MID, payload types).

Bug: webrtc:4050
Change-Id: I16b68481d01f921c326a33f5f2baf79d8b3f12e2
Reviewed-on: https://chromium-review.googlesource.com/590762
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19396}
2017-08-17 23:01:59 +00:00
..
api
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
2017-08-17 15:58:54 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
call
Rename RsidResolutionObserver to SsrcBindingObserver.
2017-08-17 23:01:59 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ )
2017-08-15 19:31:23 +00:00
examples
Remove dep on system_wrappers for apprtc and use only framework.
2017-08-16 14:28:03 +00:00
logging
Add rtc_event_log_unittest_helper.h to relevant BUILD.gn
2017-08-17 15:32:39 +00:00
media
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
2017-08-17 15:58:54 +00:00
modules
Share ScreenDrawerLockPosix implementation
2017-08-17 20:47:38 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Refactor TurnPort tests.
2017-08-17 21:58:09 +00:00
pc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
rtc_base
Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )
2017-08-17 14:59:54 +00:00
rtc_tools
Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.
2017-08-16 09:48:33 +00:00
sdk
Fix a reference leak when converting I420 VideoFrame to I420Frame.
2017-08-17 10:56:58 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names
2017-08-17 12:31:02 +00:00
video
Revert of quest keyframes more frequently on stream start/decoding error. (patchset #2 id:170001 of https://codereview.webrtc.org/2996823002/ )
2017-08-17 18:01:46 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +00:00
.gitignore
…
BUILD.gn
Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )
2017-08-17 14:59:54 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
whitespace.txt
…

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.