platform-external-webrtc

Files

jmarusic@webrtc.org 0cb612b43b We changed Encode() and EncodeInternal() return type from bool to void in this issue:

https://webrtc-codereview.appspot.com/38279004/
Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43839004

Cr-Commit-Position: refs/heads/master@{#8749}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-03-17 12:13:13 +00:00

base

Chromium build fix: Include new .cc files in rtc_base

2015-03-10 01:00:17 +00:00

build

Remove libvpx suppression due to fixed bug.

2015-03-16 20:43:47 +00:00

common_audio

Clean up LappedTransform and Blocker.

2015-03-12 23:24:19 +00:00

common_video

Add I420 buffer pool to avoid unnecessary allocations

2015-03-17 11:41:15 +00:00

examples

Roll chromium_revision 00e438c..8d51d96 (320241:320682)

2015-03-16 09:00:41 +00:00

libjingle

Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away

2015-02-26 14:43:50 +00:00

modules

We changed Encode() and EncodeInternal() return type from bool to void in this issue:

2015-03-17 12:13:13 +00:00

overrides

Re-landing perf improvement for libjingle logging after reverting the general change.

2015-03-07 12:18:14 +00:00

p2p

Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.

2015-03-16 21:16:23 +00:00

sound

Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.

2015-02-12 11:55:32 +00:00

system_wrappers

Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead.

2015-03-16 16:06:16 +00:00

test

Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."

2015-03-16 13:48:18 +00:00

tools

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

video

Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."

2015-03-16 13:48:18 +00:00

video_engine

2015-03-16 16:06:16 +00:00

voice_engine

Supporting Opus DTX in Voice Engine.

2015-03-13 09:38:55 +00:00

.gitignore

…

BUILD.gn

Let Chromium declare the mips_dsp_rev build variable.

2015-03-04 09:51:17 +00:00

call.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.cc

Remove DCHECK from common_types.cc

2015-03-04 20:10:38 +00:00

common_types.h

Add CVO support to RTP sender side.

2015-03-04 22:55:43 +00:00

common.gyp

Fix style violations in common_types.h and config.h

2015-02-26 14:01:28 +00:00

common.h

…

config.cc

Fix style violations in common_types.h and config.h

2015-02-26 14:01:28 +00:00

config.h

Add CVO support to Vie layer.

2015-03-12 20:51:50 +00:00

engine_configurations.h

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

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

frame_callback.h

Changing include guard in frame_callback.h.

2015-02-03 14:51:39 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Remove webrtc/system_wrappers/interface/scoped_ptr.h

2015-03-03 10:17:49 +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

…

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

supplement.gypi

Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)

2015-02-18 10:38:11 +00:00

transport.h

…

typedefs.h

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

video_decoder.h

Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."

2015-03-16 13:48:18 +00:00

video_encoder.h

Remove default arguments in EncodedImageCallback.

2015-02-09 09:14:48 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

video_frame.h

I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.

2015-03-16 13:26:41 +00:00

video_receive_stream.h

Add decoder-timing stats to VideoReceiveStream.

2015-02-25 10:42:45 +00:00

video_renderer.h

Use VideoReceiveStream as an ExternalRenderer.

2015-02-09 15:15:24 +00:00

video_send_stream.h

Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."

2015-03-10 15:13:13 +00:00

webrtc_examples.gyp

Improve cleaning for Android demo applications

2015-02-28 11:18:20 +00:00

webrtc_perf_tests.isolate

Offline screenshare quality test, plus loopback.

2015-02-18 12:46:44 +00:00

webrtc_tests.gypi

rtc_unittests on Android

2015-03-04 08:48:17 +00:00

webrtc.gyp

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +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.