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brandtr 4553562fe3 Remove unnecessary RTPFragmentationHeader from VideoProcessor callbacks. 
This test is and should be independent of RTP, so we don't need the
information provided in this struct.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2995403002
Cr-Commit-Position: refs/heads/master@{#19443}
2017-08-22 10:33:11 +00:00
..
api
iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs
2017-08-21 13:11:18 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
call
Fix size_t to int in RtpDemuxer.
2017-08-21 20:22:54 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Change logging method from WEBRTC_TRACE macro to LOG macro.
2017-08-22 07:29:13 +00:00
examples
Remove dep on system_wrappers for apprtc and use only framework.
2017-08-16 14:28:03 +00:00
logging
Add rtc_event_log_unittest_helper.h to relevant BUILD.gn
2017-08-17 15:32:39 +00:00
media
Revert of Modify profiles for H264 encode SW fallback (patchset #2 id:20001 of https://codereview.webrtc.org/2997913003/ )
2017-08-22 00:52:41 +00:00
modules
Remove unnecessary RTPFragmentationHeader from VideoProcessor callbacks.
2017-08-22 10:33:11 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Add TLS TURN tests.
2017-08-17 23:03:04 +00:00
pc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
rtc_base
Fix a delete type mismatch (deleting a sockaddr_in6* as a sockaddr*)
2017-08-18 07:15:19 +00:00
rtc_tools
Delete old CongestionController class
2017-08-21 10:43:33 +00:00
sdk
Remove unnecessary release call in EglRenderer.
2017-08-22 07:21:48 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Delete old CongestionController class
2017-08-21 10:43:33 +00:00
video
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
2017-08-22 09:08:51 +00:00
voice_engine
Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ )
2017-08-18 14:22:36 +00:00
.gitignore
…
BUILD.gn
Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )
2017-08-17 14:59:54 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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_receive_stream.h
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +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.