platform-external-webrtc

Files

elad.alon cfd88bbe80 Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_

BUG=webrtc:7334

Review-Url: https://codereview.webrtc.org/2742383002
Cr-Commit-Position: refs/heads/master@{#17222}

2017-03-14 09:50:46 +00:00

api

Change the type of session_id() from string to int64_t.

2017-03-13 18:00:54 +00:00

audio

Move delay_based_bwe_ into CongestionController

2017-03-09 14:40:08 +00:00

base

Refactor Windows TaskQueue code to only need a single high res timer.

2017-03-10 17:33:53 +00:00

call

Revert of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #2 id:90001 of https://codereview.webrtc.org/2744003002/ )

2017-03-13 08:54:13 +00:00

common_audio

iSAC fix entropy coder: Recently added DCHECK could in fact trigger

2017-03-13 12:28:47 +00:00

common_video

Add FullStack test for simulcast screenshare mode.

2017-03-09 17:23:30 +00:00

examples

Revert of PyLint fixes for tools-webrtc and webrtc/tools (patchset #3 id:40001 of https://codereview.webrtc.org/2736233003/ )

2017-03-09 14:09:33 +00:00

logging

Enable GN check for webrtc/logging

2017-03-08 10:12:11 +00:00

media

Revert to allowing only 1 unsignaled receive stream for audio.

2017-03-13 12:46:15 +00:00

modules

Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_

2017-03-14 09:50:46 +00:00

ortc

Add the function ToRtpParameters.

2017-03-09 01:15:06 +00:00

p2p

Making candidate pool size behave as decided in JSEP.

2017-03-10 23:18:00 +00:00

Making candidate pool size behave as decided in JSEP.

2017-03-10 23:18:00 +00:00

sdk

Trigger framelisteners even on frames dropped by the FPS reduction by default.

2017-03-13 12:11:48 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Delete utf_util_win.h.

2017-03-14 08:32:50 +00:00

test

Delete utf_util_win.h.

2017-03-14 08:32:50 +00:00

tools

Unify the FillAudioEncoderTimeSeries with existing processing functions.

2017-03-13 12:24:05 +00:00

video

Avoid holding lock while calling stream_resetter_ in MaxPaddingSetTest

2017-03-13 12:15:14 +00:00

voice_engine

Remove VoENetEqStats interface.

2017-03-13 09:36:19 +00:00

.gitignore

…

BUILD.gn

Remove voe_cmd_test.

2017-03-07 09:43:21 +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

Speculative fix for division by zero in Vp8EncoderImpl.

2017-03-10 14:08:26 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +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

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +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

Add |protected_by_flexfec| flag to VideoReceiveStream::Config.

2017-03-09 14:13:20 +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 Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )

2017-03-04 03:41:59 +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.