zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
8659d14bff902f71a2c6884dcf8bfd03363a06d5
platform-external-webrtc/webrtc
History
Haibo Huang 715411c11b Remove dependency on std::tr1 
This change is to make webrtc compatible with new googletest.

Test: compile
Change-Id: I91fa5ce8a986e35209986bef5dc0b2202155623f
2018-07-11 05:04:46 +00:00
..
api
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
audio
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
base
Convert to Android.bp
2017-10-25 15:20:25 -07:00
build
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
call
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
common_audio
Convert to Android.bp
2017-10-25 15:20:25 -07:00
common_video
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
examples
Remove dead symlink.
2017-12-15 11:01:06 -08:00
libjingle
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
modules
Remove dependency on std::tr1
2018-07-11 05:04:46 +00:00
p2p
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
sound
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
system_wrappers
Add 'vendor: true' to headers to fix vndk builds
2017-10-26 00:35:01 -07:00
test
Convert to Android.bp
2017-10-25 15:20:25 -07:00
tools
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
video
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
voice_engine
Leave only an empty top level OWNERS file.
2017-02-23 22:25:35 -08:00
.gitignore
Merge upstream SHA 04cb763
2016-01-15 11:28:47 -08:00
audio_receive_stream.h
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
…
common_types.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common.gyp
…
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.