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minyue 930ff23261 Fixing a unittest to allow new Opus version. 
New Opus version starts to support 120ms frame lengths. AudioEncoderOpusTest had an unnecessary check on the available frame lengths.

It is removed in this CL.

BUG=b/35415318

Review-Url: https://codereview.webrtc.org/2731583003
Cr-Commit-Position: refs/heads/master@{#17011}
2017-03-03 15:34:52 +00:00
..
api
Revert of GN: Include webrtc/api targets even if rtc_include_tests=false (patchset #2 id:20001 of https://codereview.webrtc.org/2725053008/ )
2017-03-03 06:26:23 +00:00
audio
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
base
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
call
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
common_audio
Enable GN check for webrtc/common_audio
2017-03-01 15:07:10 +00:00
common_video
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
examples
Add Metal video view in AppRTCMobile and Metal availability macro.
2017-03-03 14:11:10 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
modules
Fixing a unittest to allow new Opus version.
2017-03-03 15:34:52 +00:00
ortc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
p2p
Relanding: Remove the IceTransportInternal2.
2017-03-02 06:47:18 +00:00
pc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
sdk
Add Metal video view in AppRTCMobile and Metal availability macro.
2017-03-03 14:11:10 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Provide an inline implementation of IsEnabled.
2017-02-28 17:26:22 +00:00
test
Revert of fixed VP8 simulcast to not decode non-selected streams (patchset #5 id:80001 of https://codereview.webrtc.org/2728553003/ )
2017-03-02 12:59:33 +00:00
tools
Add probe logging to RtcEventLog.
2017-02-27 10:18:46 +00:00
video
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
voice_engine
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
.gitignore
…
BUILD.gn
Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )
2017-02-26 15:12:50 +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
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +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.