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webrtc_m130/webrtc
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kjellander@webrtc.org 71d9721bdc iOS: Fix JSON for tryserver configurations. 
Use boolean values instead of strings. With this landed
https://codereview.chromium.org/1723033003/ can be reverted.

BUG=498746
R=smut@google.com

Review URL: https://codereview.webrtc.org/1728743002 .

Cr-Commit-Position: refs/heads/master@{#11724}
2016-02-23 20:16:04 +00:00
..
api
Tune QP thresholds for HW H.264 encoder.
2016-02-22 23:05:26 +00:00
audio
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
base
Reduce the scope of rtc::Event::Wait() locking.
2016-02-22 10:32:02 +00:00
build
iOS: Fix JSON for tryserver configurations.
2016-02-23 20:16:04 +00:00
call
Disabled flaky tests
2016-02-23 17:56:49 +00:00
common_audio
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
2016-02-23 09:26:52 +00:00
common_video
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
examples
Add more camera resolutions to camera scaling slider.
2016-02-23 00:48:36 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
modules
Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes.
2016-02-23 16:57:54 +00:00
p2p
Remove GetTransport() from TransportChannelImpl
2016-02-20 04:43:49 +00:00
pc
Remove conference_mode flag from AudioOptions and VideoOptions.
2016-02-17 13:25:40 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Simple RTCP receiver fuzzer.
2016-02-23 10:35:41 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
Only average positive quality stats.
2016-02-23 16:30:55 +00:00
voice_engine
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
2016-02-17 18:04:26 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
…
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common.h
Added boilerplate code for being able to test the upcoming
2016-02-17 09:11:24 +00:00
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_media_unittests.isolate
Rename libjingle_media_unittest -> rtc_media_unittests
2016-02-12 14:06:11 +00:00
rtc_pc_unittests.isolate
Rename libjingle_p2p_unittest -> rtc_pc_unittests
2016-02-15 08:36:44 +00:00
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
2016-02-18 19:30:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Remove no-op VideoDecoder::Reset implementation.
2016-02-12 14:16:31 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Fix Linux 32-bit compilation after sysroot switch.
2016-02-18 20:52:32 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
webrtc.gyp
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +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.