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platform-external-webrtc/webrtc
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charujain 0f01c7f930 Added tool for reference less video analysis (go/refless-video-analysis) 
This tool takes list of video file names as input and calculates freezing metrics score for the video files without having reference to original video by comparing the PSNR and SSIM values of current and previous frame.

BUG=webrtc:6759

Review-Url: https://codereview.webrtc.org/2515253004
Cr-Commit-Position: refs/heads/master@{#15386}
2016-12-02 13:00:10 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
New gn target video_frame_api.
2016-12-02 12:01:21 +00:00
audio
Reland "Update rtt on audio only calls".
2016-11-30 15:51:19 +00:00
base
Templatize percentile_filter.h and move it to base/analytics.
2016-11-30 14:52:06 +00:00
build
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
call
Reland "Update rtt on audio only calls".
2016-11-30 15:51:19 +00:00
common_audio
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
media
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
modules
fix coding and documentary ambiguity in AimdRateControl::TimeToReduceFurther.
2016-12-02 11:27:18 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Relaxing timeouts for TestMediaMonitor.
2016-12-01 20:49:25 +00:00
sdk
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add linearly spaced counting histograms
2016-12-01 17:58:53 +00:00
test
Remove xdisplaycheck
2016-12-02 11:30:51 +00:00
tools
Added tool for reference less video analysis (go/refless-video-analysis)
2016-12-02 13:00:10 +00:00
video
Wire up BitrateAllocation to be sent as RTCP TargetBitrate
2016-12-01 14:34:18 +00:00
voice_engine
Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
2016-12-02 10:40:12 +00:00
.gitignore
…
BUILD.gn
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Remove API-related #defines from voice_engine_configurations.h
2016-12-02 10:30:23 +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.