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webrtc_m130/webrtc
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zhihuang 3a334656de Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer. 
Use the attribute in MediaContentDescription to test whether Rtx is removed in the answer instead of searching the substring "rtx" in the whole answer sdp.

BUG=webrtc:4943

Review-Url: https://codereview.webrtc.org/1919523002
Cr-Commit-Position: refs/heads/master@{#12639}
2016-05-06 01:37:52 +00:00
..
api
Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer.
2016-05-06 01:37:52 +00:00
audio
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
base
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
2016-05-04 08:54:39 +00:00
build
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
2016-05-04 09:37:50 +00:00
call
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
common_audio
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
common_video
This cl deletes the class webrtc::VideoRendererCallback.
2016-05-03 08:09:17 +00:00
examples
Update CPU monitor to use moving averages.
2016-05-04 18:04:18 +00:00
libjingle
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
2016-05-04 08:54:39 +00:00
media
Support RtpEncodingParameters::active in voice engine.
2016-05-03 20:50:24 +00:00
modules
Removed the file echo_cancellation_internal.h and moved
2016-05-05 20:34:35 +00:00
p2p
Cache a ClientHello received before the DTLS handshake has started.
2016-05-05 00:16:39 +00:00
pc
Accept all the media profiles required by JSEP.
2016-05-05 18:44:44 +00:00
sdk
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
test
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Removed SSRC knowledge from ViEEncoder.
2016-05-04 18:26:56 +00:00
voice_engine
Move ADM Create() method to public interface.
2016-05-03 19:51:31 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
2016-05-04 09:37:50 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
…
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
…
config.h
…
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
2016-05-04 08:54:39 +00:00
webrtc.gyp
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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.