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nisse 6d4dd593a8 Always call RemoteBitrateEstimator::IncomingPacket from Call. 
Delete the calls from RtpStreamReceiver (for video) and
AudioReceiveStream.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2659563002
Cr-Commit-Position: refs/heads/master@{#16393}
2017-02-01 11:06:58 +00:00
..
api
Enable cpplint and fix cpplint errors in webrtc/api
2017-02-01 09:55:59 +00:00
audio
Always call RemoteBitrateEstimator::IncomingPacket from Call.
2017-02-01 11:06:58 +00:00
base
Use correct calling convention for CreateThread callback on Windows.
2017-01-31 23:10:44 +00:00
call
Always call RemoteBitrateEstimator::IncomingPacket from Call.
2017-02-01 11:06:58 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Delete VideoFrame::set_render_time_ms.
2017-01-30 10:43:18 +00:00
examples
Reland of move usage of deprecated g_type_init API (patchset #1 id:1 of https://codereview.webrtc.org/2666103002/ )
2017-01-31 14:50:14 +00:00
logging
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
2017-01-27 12:53:07 +00:00
media
Drop frames until specified bitrate is achieved.
2017-02-01 09:31:52 +00:00
modules
Drop frames until specified bitrate is achieved.
2017-02-01 09:31:52 +00:00
p2p
Delete unneeded includes of base/common.h.
2017-01-25 09:47:24 +00:00
pc
Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods.
2017-01-31 11:34:01 +00:00
sdk
Enable Android H264 High profile decoder
2017-01-27 20:24:24 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Refactor FakeAudioDevice to have separate methods for starting recording and playout.
2017-01-31 21:32:49 +00:00
tools
Compute packet loss for event log visualization similar to how it is defined in RFC 3550.
2017-01-30 16:44:51 +00:00
video
Always call RemoteBitrateEstimator::IncomingPacket from Call.
2017-02-01 11:06:58 +00:00
voice_engine
Stop using VoEVideoSync in Call/VideoReceiveStream.
2017-01-31 11:58:40 +00:00
.gitignore
…
BUILD.gn
Move Android tests under sdk/android.
2017-01-27 13:14:59 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
2017-01-27 12:53:07 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +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.