webrtc

History

pthatcher@webrtc.org f050791ba0 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

This reverts r7992.

It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-16 22:28:03 +00:00

base

Revert "Add adapter_type into Candidate object."

2014-12-16 05:28:10 +00:00

build

Move isolate path into webrtc/build/android/test_runner.py

2014-12-11 11:59:46 +00:00

common_audio

Add (safe) uint32_t cast to fix Win64 build.

2014-12-16 20:47:42 +00:00

common_video

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

examples

Update Android projects to API level 21.

2014-10-31 23:26:10 +00:00

libjingle

Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

2014-12-16 22:28:03 +00:00

modules

Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder

2014-12-16 21:04:55 +00:00

overrides

webrtc/overrides: add OWNERS-file.

2014-09-17 08:04:28 +00:00

p2p

Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

2014-12-16 22:28:03 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Add video send bitrates to histogram stats:

2014-12-09 09:47:53 +00:00

test

Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

2014-12-15 22:09:40 +00:00

tools

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +00:00

video

Wire up RTT statistics to webrtc::Call.

2014-12-11 13:26:09 +00:00

video_engine

Add field to counters for when first rtp/rtcp packet is sent/received.

2014-12-16 12:03:11 +00:00

voice_engine

Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

2014-12-15 22:09:40 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

BUILD.gn

Enabling building with NEON on ARM64

2014-11-26 17:01:40 +00:00

call.h

Wire up RTT statistics to webrtc::Call.

2014-12-11 13:26:09 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

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

common_types.h

Add field to counters for when first rtp/rtcp packet is sent/received.

2014-12-16 12:03:11 +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

Log formatting fix for VideoEncoderConfig.

2014-11-06 09:35:08 +00:00

config.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

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

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

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

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

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_receive_stream.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +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

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

webrtc_examples.gyp

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

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

webrtc_tests.gypi

Add AGC manager tests.

2014-12-16 14:48:47 +00:00

webrtc.gyp

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +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.