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webrtc_m130/webrtc
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perkj d61bf803d2 Removed MediaStreamTrackInterface::set_state 
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.

Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1816143002

Cr-Commit-Position: refs/heads/master@{#12115}
2016-03-24 10:16:23 +00:00
..
api
Removed MediaStreamTrackInterface::set_state
2016-03-24 10:16:23 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Move swap_queue.h to base/
2016-03-24 08:51:59 +00:00
build
Add visibility flag to GYP.
2016-03-23 20:19:23 +00:00
call
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
common_audio
Move swap_queue.h to base/
2016-03-24 08:51:59 +00:00
common_video
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
examples
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
libjingle
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
media
Delete default_send_ssrc_.
2016-03-24 08:02:55 +00:00
modules
Reland of Added a bitexactness test for the echo canceller in the audio processing module.
2016-03-24 09:45:58 +00:00
p2p
Add 16-bit network id to the candidate signaling.
2016-03-23 23:07:54 +00:00
pc
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-20 13:15:48 +00:00
system_wrappers
Fix race condition in EventTimerPosix
2016-03-22 08:51:46 +00:00
test
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
voice_engine
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +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
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +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
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
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
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 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
webrtc_examples.gyp
Update iOS AppRTCDemo to use the updated Objective-C API.
2016-03-14 05:08:35 +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
Remove webrtc/sound/ subdir.
2016-03-17 18:23:05 +00:00
webrtc.gyp
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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.