platform-external-webrtc

Files

zijiehe 8fefe9889d [DesktopCapturer] FallbackDesktopCapturerWrapper and its tests

FallbackDesktopCapturerWrapper is a DesktopCapturer implementation, which owns
two DesktopCapturer implementations. If the main DesktopCapturer fails, it uses
the secondary capturer. The logic is now used in ScreenCapturerWinMagnifier, and
it can also be shared in ScreenCapturerWinDirectx to fallback to Gdi capturer on
privilege prompt or login screen.

BUG=684937

Review-Url: https://codereview.webrtc.org/2697453002
Cr-Commit-Position: refs/heads/master@{#16677}

2017-02-17 22:32:04 +00:00

api

Adding RTCErrorOr class to be used by ORTC APIs.

2017-02-17 07:31:33 +00:00

audio

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

base

Don't use CONF_VALUE in VerifyServerName.

2017-02-17 21:04:43 +00:00

call

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-02-09 11:05:59 +00:00

common_video

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

examples

Modify android video_quality_loopback_test to run commands from the src dir.

2017-02-16 15:20:26 +00:00

logging

Delete class SSRCDatabase, and its global ssrc registry,

2017-02-17 16:34:35 +00:00

media

Be less pessimistic about turning "default" receive streams into signaled streams.

2017-02-17 20:01:14 +00:00

modules

[DesktopCapturer] FallbackDesktopCapturerWrapper and its tests

2017-02-17 22:32:04 +00:00

p2p

Use rtc::ToString instead of std::to_string.

2017-02-16 19:10:51 +00:00

Adding RTCErrorOr class to be used by ORTC APIs.

2017-02-17 07:31:33 +00:00

sdk

Relanding: Add the url attribute to the IceCandidate (Java Wrapper)

2017-02-17 20:45:00 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

2017-01-27 14:35:16 +00:00

system_wrappers

Change NtpTime representation to single uint64_t

2017-02-15 09:18:15 +00:00

test

Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.

2017-02-16 14:52:32 +00:00

tools

Propagate packet pacing information to SenTimeHistory.

2017-02-17 11:59:43 +00:00

video

Replace DCHECK with ASSERT_TRUE in vie_encoder_unittest.cc

2017-02-16 23:46:27 +00:00

voice_engine

Delete class SSRCDatabase, and its global ssrc registry,

2017-02-17 16:34:35 +00:00

.gitignore

…

BUILD.gn

Adding RTCErrorOr class to be used by ORTC APIs.

2017-02-17 07:31:33 +00:00

call.h

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

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

codereview.settings

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common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Remove VoEExternalMedia interface.

2017-02-14 12:55:00 +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

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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

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PRESUBMIT.py

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README.chromium

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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

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_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

webrtc.gni

Move iOS tests to XCTest from gtest.

2017-02-14 12:58:56 +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.