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Danil Chapovalov 84b4d2c1c2 Use rtp_header_extension_map.h instead of rtp_header_extension.h 
Finish renaming started in the https://chromium-review.googlesource.com/c/520947/

Bug: webrtc:5565
Change-Id: If420e05165ef7c110b7d38f53dbe73c21a4059bc
Reviewed-on: https://chromium-review.googlesource.com/528095
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18538}
2017-06-12 14:01:20 +00:00
..
api
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
2017-06-02 21:37:37 +00:00
audio
New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.
2017-06-01 11:02:35 +00:00
base
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
2017-06-12 12:53:35 +00:00
call
Use rtp_header_extension_map.h instead of rtp_header_extension.h
2017-06-12 14:01:20 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Update webrtc/test to new VideoFrameBuffer interface
2017-06-11 16:10:16 +00:00
examples
List all device resolutions in AppRTCMobile settings
2017-06-12 10:02:44 +00:00
logging
Use rtp_header_extension_map.h instead of rtp_header_extension.h
2017-06-12 14:01:20 +00:00
media
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
modules
Use rtp_header_extension_map.h instead of rtp_header_extension.h
2017-06-12 14:01:20 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Fix issue with send-side bandwidth estimation over TURN TCP connections.
2017-05-27 01:40:05 +00:00
pc
s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
2017-06-12 08:16:46 +00:00
sdk
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
2017-06-12 12:53:35 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Use rtp_header_extension_map.h instead of rtp_header_extension.h
2017-06-12 14:01:20 +00:00
tools
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
2017-06-12 12:53:35 +00:00
video
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
2017-06-12 12:53:35 +00:00
voice_engine
Fix Channel::GetSendCodec when used together with SetEncoder.
2017-06-09 14:30:13 +00:00
.gitignore
…
BUILD.gn
Split rtc_task_queue target. Add separate target for sequenced_task_checker and weak_ptr.
2017-06-12 12:53:35 +00:00
codereview.settings
…
common_types.cc
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
common_types.h
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +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
Delete webrtc/call.h (replaced with webrtc/call/call.h).
2017-06-08 09:21:17 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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 Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.