platform-external-webrtc

Files

henrik.lundin@webrtc.org 18617cfde8 Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined

This requires two fixes:
1. Use DCHECK instead of assert in ThreadChecker's unittest.

2. Activate DCHECK when DCHECK_ALWAYS_ON in enabled.

Both these modifications are in line with Chromium's implementation.
The ThreadChecker unittest was changed to use assert instead of DCHECK
on the initial import (since WebRTC did not have a DCHECK back then).

BUG=3803
TEST=local out/{Debug,Release}/rtc_unittests built with and without DCHECK_ALWAYS_ON
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7178 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-15 11:19:35 +00:00

base

Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined

2014-09-15 11:19:35 +00:00

build

Set minimum SDK level to 10.7 for Mac and iOS.

2014-09-15 08:02:43 +00:00

common_audio

Add ctors to ChannelBuffer to enable copying on construction.

2014-09-08 20:27:04 +00:00

common_video

Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."

2014-09-11 09:48:30 +00:00

examples

Fix compile error on JDK 1.7.

2014-09-10 12:35:59 +00:00

libjingle/xmllite

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

modules

Add enable flag for Android device orientation change event.

2014-09-12 16:48:12 +00:00

overrides/webrtc/base

Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.

2014-05-21 16:52:14 +00:00

sound

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

system_wrappers

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

test

Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."

2014-09-11 09:48:30 +00:00

tools

Fix printing of error stack in rtcbot when a test fails via test.fail().

2014-09-10 14:35:35 +00:00

video

Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."

2014-09-11 09:48:30 +00:00

video_engine

Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.

2014-09-12 07:42:33 +00:00

voice_engine

Calculating round-trip-time in send-only channel in VoE.

2014-09-11 07:51:53 +00:00

.gitignore

…

BUILD.gn

Fix the RTC+Chromium GN build.

2014-09-09 19:15:33 +00:00

call.h

Network up/down signaling in Call.

2014-09-03 16:17:12 +00:00

common_types.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

config.h

Check before send/receive rtp header extensions.

2014-07-20 15:27:35 +00:00

engine_configurations.h

Add boilerplate code for H.264.

2014-07-04 12:42:07 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

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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

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supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Add CHECK and friends from Chromium.

2014-08-28 16:28:26 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

video_receive_stream.h

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

webrtc_examples.gyp

Fix rm command for class cleanup in r7091

2014-09-06 22:11:28 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

webrtc_tests.gypi

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

webrtc.gyp

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +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.