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platform-external-webrtc/webrtc
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aleloi 2ae5fdff86 Made MixerAudioSource a pure interface. 
This required quite a few small changes in the mixing algorithm
structure, the mixer interface and the mixer unit tests.

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2396483002
Cr-Commit-Position: refs/heads/master@{#14567}
2016-10-07 11:30:19 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCIceCandidateStats[1] added.
2016-10-07 09:18:55 +00:00
audio
Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
2016-10-04 09:45:54 +00:00
base
Add support for component builds on Windows Swarming. Attempt 2
2016-10-07 09:47:45 +00:00
build
Add iOS static library GN build script.
2016-10-05 20:16:07 +00:00
call
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
common_audio
Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere
2016-10-04 20:47:02 +00:00
common_video
Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )
2016-10-06 13:29:34 +00:00
examples
Speculatively increase timeouts for DirectRTCClientTest to reduce flakiness.
2016-10-06 07:34:47 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
logging
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
media
H264 codec: Check profile-level-id when matching
2016-10-06 09:15:56 +00:00
modules
Made MixerAudioSource a pure interface.
2016-10-07 11:30:19 +00:00
p2p
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
pc
Delete unused class cricket::MediaSinkInterface, and mediasink.h.
2016-10-04 12:33:53 +00:00
sdk
GN: Add missing config //webrtc:common_objc for rtc_sdk_common_objc
2016-10-07 05:18:16 +00:00
stats
RTCIceCandidateStats[1] added.
2016-10-07 09:18:55 +00:00
system_wrappers
Fixed flaky clock_unittest by using relative comparison.
2016-10-05 08:55:10 +00:00
test
Harmonize using-declarations in FEC header formatter fuzzers.
2016-10-05 14:35:01 +00:00
tools
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
video
Cleanup unused dependency on video_capture_module.
2016-10-07 07:13:26 +00:00
voice_engine
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
.gitignore
…
BUILD.gn
Make bwe_simulator a separate test target that doesnt run on the bots
2016-10-07 07:40:03 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
config.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Make MediaCodecEncoder fallback to a software encoder on failure.
2016-10-06 12:55:19 +00:00
video_frame.h
Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )
2016-10-06 13:29:34 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.