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webrtc_m130/webrtc
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glaznev 489c0d4832 Decrease threshold for key frame generation. 
On some recent Android devices camera switch is completed in 400 ms.
Need to adjust key frame generation threshold to ensure HW encoder
still generates a key frame after camera switch to workaround video
distortions.

BUG=b/32238476

Review-Url: https://codereview.webrtc.org/2447163003
Cr-Commit-Position: refs/heads/master@{#14791}
2016-10-26 17:53:05 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Decrease threshold for key frame generation.
2016-10-26 17:53:05 +00:00
audio
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
base
Disable TaskQueueTest.PostDelayed because of flakiness
2016-10-26 16:04:35 +00:00
build
MB: Add Linux swarming bots with memory sanitizers on the FYI waterfall.
2016-10-26 14:00:29 +00:00
call
Fix BWE simulations so that it uses the delay based BWE.
2016-10-25 14:04:44 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
2016-10-26 09:48:24 +00:00
examples
Testing of VideoFileRenderer with byte frames
2016-10-26 08:47:14 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
media
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
modules
- Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing.
2016-10-26 12:12:29 +00:00
p2p
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn
2016-10-24 09:09:40 +00:00
tools
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
2016-10-18 08:44:50 +00:00
video
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
voice_engine
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +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
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +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
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.