zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
09baefeaf9f5f80d078b32afa9656ebdb1069c39
platform-external-webrtc/webrtc
History
charujain 09baefeaf9 GN: Remove non-existing header file: media/base/videorenderer.h 
This header was removed in https://codereview.webrtc.org/1819103003
but the BUILD.gn file was not updated.

BUG=webrtc:5949
NOTRY=True
TBR=nisse@webrtc.org

Review-Url: https://codereview.webrtc.org/2359113002
Cr-Commit-Position: refs/heads/master@{#14349}
2016-09-22 11:51:57 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
New class AdaptedVideoTrackSource.
2016-09-22 08:26:06 +00:00
audio
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
base
Revert of Allow the DTLS fingerprint verification to occur after the handshake. (patchset #11 id:200001 of https://codereview.webrtc.org/2163683003/ )
2016-09-20 00:21:00 +00:00
build
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
call
Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets.
2016-09-21 13:51:52 +00:00
common_audio
Compensate for the IntelligibilityEnhancer processing delay in high bands
2016-09-20 21:52:08 +00:00
common_video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
examples
Android VideoSource: Add adaptOutputFormat function
2016-09-21 14:20:16 +00:00
libjingle
GN: Change group deps to public_deps.
2016-09-15 07:53:34 +00:00
media
GN: Remove non-existing header file: media/base/videorenderer.h
2016-09-22 11:51:57 +00:00
modules
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
p2p
Remove dead code branches from P2PtransportChannel unittest.
2016-09-20 07:23:33 +00:00
pc
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
sdk
GN: Enable rtc_common_config for more targets.
2016-09-18 15:12:36 +00:00
stats
RTCTestStats moved into webrtc/stats/test (rtc_stats_test_utils)
2016-09-19 13:06:05 +00:00
system_wrappers
Roll chromium_revision cf9457edb7..cede888c27 (416297:419407)
2016-09-20 20:23:12 +00:00
test
Generalize FEC unit tests and rename GenerateFec.
2016-09-21 06:16:36 +00:00
tools
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
video
Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
2016-09-21 10:52:21 +00:00
voice_engine
Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
2016-09-22 10:36:34 +00:00
.gitignore
…
BUILD.gn
Reenabled the RtcEventLog unittests
2016-09-19 21:42:10 +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
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
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
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +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
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Ignores warning for SecRandomCopyBytes() using Xcode 8
2016-09-21 11:13:07 +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
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.