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jianj 20acdf2443 Add vp9 QP parser. 
BUG=webrtc:7662

Review-Url: https://codereview.webrtc.org/2891803003
Cr-Commit-Position: refs/heads/master@{#18260}
2017-05-24 17:00:16 +00:00
..
api
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
audio
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +00:00
base
Revert of Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations. (patchset #3 id:40001 of https://codereview.webrtc.org/2885143006/ )
2017-05-23 19:55:03 +00:00
call
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
examples
AppRTCMobile support to turn off the WebRTC AGC and HPF
2017-05-18 22:09:33 +00:00
logging
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
media
Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
2017-05-24 11:19:54 +00:00
modules
Add vp9 QP parser.
2017-05-24 17:00:16 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
pc
Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ )
2017-05-23 14:34:17 +00:00
sdk
Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ )
2017-05-23 14:34:17 +00:00
stats
…
system_wrappers
Adding target to track asm_defines.h
2017-05-15 21:59:17 +00:00
test
Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ )
2017-05-23 14:34:17 +00:00
tools
Revert of Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric n… (patchset #1 id:1 of https://codereview.webrtc.org/2901793002/ )
2017-05-23 15:31:11 +00:00
video
Fix packetization logic to leave space for extensions in the last packet
2017-05-23 16:34:21 +00:00
voice_engine
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
.gitignore
…
BUILD.gn
Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )
2017-05-17 13:52:58 +00:00
call.h
…
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_frame.h
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
webrtc.gni
Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ )
2017-05-23 14:34:17 +00:00
whitespace.txt
…

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.