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platform-external-webrtc/webrtc
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jackychen afaae0d151 External VNR speed improvement. 
Improved visual quality with 3x times speed-up.
Change list:
 1. Remove second chance filter in temporal denoising filter to mitigate trailing artifact.
 2. Add swap buffer to save one whole-frame memcpy.
 3. Do noise estimation on every N blocks.
 4. Adopt a faster moving object detection algorithm (change the structure).
 5. Refactor the for loops and PositionCheck().
 6. Refactor the function ReduceFalseDetection (RFD).
 7. Fix a bug in TrailingBlock() which causes a mismatch.
 8. Change unit test to support swap buffer test.
 9. Remove CopyMem8x8, use memcpy to copy U/V plane which can be optimized future.
 10. Remove DenoiseMetrics.

Review URL: https://codereview.webrtc.org/1871853003

Cr-Commit-Position: refs/heads/master@{#12340}
2016-04-13 06:03:11 +00:00
..
api
Enable H.264 HW decoder soft rest.
2016-04-12 23:39:50 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Add WriteUVarint to ByteBufferWriter and ReadUVarint to ByteBufferReader
2016-04-11 23:11:47 +00:00
build
Fix paths to Android tests .isolate files.
2016-04-12 20:15:18 +00:00
call
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
common_audio
Moved struct definition to the header file for the ring buffer.
2016-04-11 05:51:10 +00:00
common_video
Add CoreVideoFrameBuffer.
2016-04-04 21:10:47 +00:00
examples
Add test annotations to AppRTCDemoTest.
2016-04-12 04:27:40 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Generate FMTP parameters for the H.264 codec.
2016-04-12 17:29:20 +00:00
modules
External VNR speed improvement.
2016-04-13 06:03:11 +00:00
p2p
Add QuicTransportChannel methods for QUIC streams
2016-04-11 23:56:26 +00:00
pc
Accept all the media profiles required by JSEP.
2016-04-13 01:32:36 +00:00
system_wrappers
Implement CPU feature detection for ARM Linux.
2016-04-06 21:02:34 +00:00
test
Fixed rtcp rpsi parsing of invalid packets.
2016-04-11 17:05:06 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Add missing tracing to RtpSender objects.
2016-04-11 09:45:43 +00:00
voice_engine
Switch to using new ACM methods for encoder management
2016-04-06 19:22:45 +00:00
.gitignore
…
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +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
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
webrtc.gyp
GYP: Add webrtc/pc/pc.gyp:* to 'All' target.
2016-04-01 09:15:05 +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.