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Henrik Lundin 82d6f2a3f7 ACM: Remove ACMVQMonCallback object 
It was never used, and the underlying functionality was removed long
ago.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1365193003 .

Cr-Commit-Position: refs/heads/master@{#10083}
2015-09-28 08:25:33 +00:00
..
audio
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
base
Disable TestUdpReadyToSendIPv4 under MSan.
2015-09-25 18:00:54 +00:00
build
Update isolate.gypi to support Swarming + move .isolate files
2015-09-25 20:19:21 +00:00
call
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Android AppRTCDemo: Add slider for changing camera capture quality during call
2015-09-25 06:23:49 +00:00
libjingle
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
modules
ACM: Remove ACMVQMonCallback object
2015-09-28 08:25:33 +00:00
overrides
Remove overridden basictypes.h.
2015-09-18 08:48:39 +00:00
p2p
Do not prune if the current best connection is weak.
2015-09-24 20:14:51 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
test
This relands "Tool to convert RtcEventLog files to RtpDump format.", commit 35624c2c3686a2ad40daffe073aa78507b0ef88e.
2015-09-18 13:41:18 +00:00
tools
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
video_engine
Reland "Wire up send-side bandwidth estimation."
2015-09-24 13:07:17 +00:00
voice_engine
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
.gitignore
…
audio_receive_stream.h
…
audio_send_stream.h
…
BUILD.gn
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
call.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
codereview.settings
…
common_types.cc
…
common_types.h
Remove channel ids from various interfaces.
2015-09-17 21:06:02 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
…
config.h
…
engine_configurations.h
Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
2015-09-25 08:36:11 +00:00
frame_callback.h
…
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
…
video_engine_tests.isolate
…
video_frame.h
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
video_receive_stream.h
…
video_renderer.h
…
video_send_stream.h
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
2015-09-08 12:13:25 +00:00
webrtc_examples.gyp
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +00:00
webrtc.gyp
Split webrtc/video into webrtc/{audio,call,video}.
2015-09-25 11:58:39 +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.