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platform-external-webrtc/webrtc
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sprang e2d83d6560 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() 
Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
2016-02-19 17:03:34 +00:00
..
api
VideoCapturerAndroid: Report onFirstFrameAvailable() for textures as well
2016-02-18 14:47:55 +00:00
audio
Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
2016-02-17 18:04:26 +00:00
base
Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ )
2016-02-19 15:16:19 +00:00
build
Compile rtc_base_objc for Mac.
2016-02-17 22:45:00 +00:00
call
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
2016-02-19 17:03:34 +00:00
common_audio
iOS: Enable modules_unittests and common_audio_unittests
2016-02-18 21:15:17 +00:00
common_video
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
examples
Replaced eglbase_jni with with holding a EglBase in PeerConnectionFactory.
2016-02-15 14:28:40 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Apply VideoOptions per stream.
2016-02-17 15:24:57 +00:00
modules
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
2016-02-19 17:03:34 +00:00
p2p
Revert of Remove GetTransport() from TransportChannelImpl (patchset #3 id:40001 of https://codereview.webrtc.org/1691673002/ )
2016-02-18 09:57:56 +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
Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
2016-02-18 19:30:25 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
2016-02-19 17:03:34 +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
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +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
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
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.