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webrtc_m130/webrtc
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Stefan Holmer 55674ffb32 Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. 
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.

R=tommi@webrtc.org
TBR=pthatcher@webtrc.org

BUG=4173

Review URL: https://codereview.webrtc.org/1589563003 .

Cr-Commit-Position: refs/heads/master@{#11251}
2016-01-14 14:49:23 +00:00
..
api
Update API for Objective-C RTCMediaSource.
2016-01-11 22:39:05 +00:00
audio
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
base
Update with new default boringssl no-aes cipher suites. Re-enable tests.
2016-01-14 14:18:02 +00:00
build
Add build_protobuf variable.
2016-01-13 13:47:07 +00:00
call
Add ramp-up tests for transport sequence number with and w/o audio.
2016-01-14 09:00:34 +00:00
common_audio
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common_video
Clean the code for external denoiser.
2016-01-13 13:36:40 +00:00
examples
Use getExternalStorageDirectory() for trace file.
2015-12-30 17:23:35 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
modules
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
p2p
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
2016-01-14 14:49:23 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
test
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
tools
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
video
Re-enable tests that failed under Linux_Msan.
2016-01-13 16:35:51 +00:00
voice_engine
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +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
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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
Add unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +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
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Step 1 to prepare call_test.* for combined audio/video tests.
2015-12-21 11:14:05 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.