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webrtc_m130/webrtc
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kthelgason 4a85abb80e Add support for more resolutions on iOS/macOS 
BUG=

Review-Url: https://codereview.webrtc.org/2231033002
Cr-Commit-Position: refs/heads/master@{#13828}
2016-08-19 08:24:49 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Adding test for unordered, fragmented SCTP message delivery.
2016-08-18 18:40:45 +00:00
audio
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
base
GN: Fix errors when some variables are set to non-default values.
2016-08-19 08:07:33 +00:00
build
Fix get_landmines.py script.
2016-08-17 11:00:47 +00:00
call
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
common_audio
WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value)
2016-08-18 10:17:10 +00:00
common_video
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
examples
Add kjellander@webrtc.org to more BUILD.gn OWNERS files.
2016-08-19 07:48:39 +00:00
libjingle
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media
GN: Fix errors when some variables are set to non-default values.
2016-08-19 08:07:33 +00:00
modules
Corrected the testvectors for the level controller
2016-08-18 16:47:52 +00:00
p2p
Do not switch to a high-cost connection that is not receiving.
2016-08-17 23:12:58 +00:00
pc
Add kjellander@webrtc.org to more BUILD.gn OWNERS files.
2016-08-19 07:48:39 +00:00
sdk
Add support for more resolutions on iOS/macOS
2016-08-19 08:24:49 +00:00
system_wrappers
GN: Update tests to have the correct shard timeout value on Android.
2016-08-15 09:29:19 +00:00
test
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
tools
Don't simulate probing based on rtc event logs since we don't have that info logged.
2016-08-17 09:14:38 +00:00
video
Reland of Adding audio to video_quality_test.
2016-08-18 13:28:59 +00:00
voice_engine
voice_engine: Removed old variants of Channel constructor and CreateChannel
2016-08-18 11:23:04 +00:00
.gitignore
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audio_receive_stream.h
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audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
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audio_state.h
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BUILD.gn
GN: Add Android support to video_engine_tests.
2016-08-18 07:42:05 +00:00
call.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
codereview.settings
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common_types.cc
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common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
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common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
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config.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
DEPS
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engine_configurations.h
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests_apk.isolate
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rtc_unittests.isolate
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supplement.gypi
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transport.h
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typedefs.h
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video_decoder.h
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video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
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video_engine_tests.isolate
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video_frame.h
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video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
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webrtc_nonparallel_tests.isolate
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webrtc_perf_tests_apk.isolate
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webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc.gyp
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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.