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nisse 1509fa1aa9 Delete cricket::VideoRenderer. 
TBR=glaznev@webrtc.org (deleting an #include in main_wnd.h)
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1819103003

Cr-Commit-Position: refs/heads/master@{#12101}
2016-03-23 11:06:05 +00:00
..
api
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
[iOS] Link with base/maccocoathreadhelper.mm on iOS.
2016-03-22 10:32:22 +00:00
build
Move build scripts to webrtc/build/ios
2016-03-21 18:38:35 +00:00
call
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
common_audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
common_video
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
examples
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
libjingle
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
media
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
modules
Remove redefined macros from BitrateAdjuster
2016-03-22 23:05:35 +00:00
p2p
Stop using some scoped_ptr features that unique_ptr doesn't have
2016-03-18 05:54:49 +00:00
pc
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
system_wrappers
Fix race condition in EventTimerPosix
2016-03-22 08:51:46 +00:00
test
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
voice_engine
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +00:00
.gitignore
…
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
…
common_types.cc
…
common_types.h
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
…
DEPS
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_media_unittests.isolate
…
rtc_pc_unittests.isolate
Rename libjingle_p2p_unittest -> rtc_pc_unittests
2016-02-15 08:36:44 +00:00
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +00:00
transport.h
…
typedefs.h
…
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_engine_tests.isolate
…
video_frame.h
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
video_renderer.h
This is an initial cleanup step, aiming to delete the
2016-03-21 08:28:06 +00:00
video_send_stream.h
Make ReconfigureVideoEncoder void.
2016-03-02 16:00:07 +00:00
webrtc_examples.gyp
Update iOS AppRTCDemo to use the updated Objective-C API.
2016-03-14 05:08:35 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Remove webrtc/sound/ subdir.
2016-03-17 18:23:05 +00:00
webrtc.gyp
Remove webrtc/sound/ subdir.
2016-03-17 18:23:05 +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.