webrtc_m130

History

nisse 4709e8971b Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

We can then drop the CongestionController and RemoteBitrateEstimator
completely from the receive streams.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2669463006
Cr-Commit-Position: refs/heads/master@{#16459}

2017-02-07 09:18:43 +00:00

api

Adding C++ versions of currently spec'd "RtpParameters" structs.

2017-02-04 20:09:01 +00:00

audio

Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

2017-02-07 09:18:43 +00:00

base

Pick the DTLS handshake timeout based on the ICE RTT estimate

2017-02-03 01:18:37 +00:00

call

Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

2017-02-07 09:18:43 +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

Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.

2017-02-06 10:21:11 +00:00

examples

Remove NewApi lint suppression.

2017-02-06 13:34:26 +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

Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily.

2017-02-06 21:03:19 +00:00

modules

Enable send-side BWE by default for video in call tests.

2017-02-06 14:29:38 +00:00

p2p

Fix Chromium FYI bot

2017-02-03 02:22:46 +00:00

Adding C++ versions of currently spec'd "RtpParameters" structs.

2017-02-04 20:09:01 +00:00

sdk

Remove NewApi lint suppression.

2017-02-06 13:34:26 +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

Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.

2017-02-06 20:53:57 +00:00

tools

Remove all occurrences of "using std::string".

2017-02-02 16:10:00 +00:00

video

Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

2017-02-07 09:18:43 +00:00

voice_engine

Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.

2017-02-06 20:53:57 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

BUILD.gn

Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.

2017-02-06 10:21:11 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

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

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

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

Remove temporary AddRtxInfo member function.

2017-02-06 10:34:25 +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

Adding build switch for Opus that supports 120ms ptime.

2017-02-02 01:31:11 +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.