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webrtc_m130/webrtc
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peah 8d13c4fe1a Changed the AEC SubbandCoherence function to not use the full AEC state 
This CL is step towards simplifying the AEC code, making it more modifiable and modular.

The changes should be bitexact.

BUG=webrtc:5201, webrtc:5298

Review-Url: https://codereview.webrtc.org/1936173002
Cr-Commit-Position: refs/heads/master@{#12652}
2016-05-07 22:03:55 +00:00
..
api
Only generate one CNAME per PeerConnection.
2016-05-07 01:40:35 +00:00
audio
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
base
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
build
Fix BoringSSL license path.
2016-05-06 18:19:54 +00:00
call
Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
2016-05-04 15:13:06 +00:00
common_audio
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
common_video
This cl deletes the class webrtc::VideoRendererCallback.
2016-05-03 08:09:17 +00:00
examples
Provide isAudioEnabled flag to control audio unit.
2016-05-07 01:54:21 +00:00
libjingle
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
media
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
modules
Changed the AEC SubbandCoherence function to not use the full AEC state
2016-05-07 22:03:55 +00:00
p2p
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
pc
Only generate one CNAME per PeerConnection.
2016-05-07 01:40:35 +00:00
sdk
Fixed a crash in Objective-C clients when data channel becomes closed.
2016-05-06 23:10:51 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
test
Corrected bug in checking the third number and added extra checks
2016-05-06 13:12:27 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
voice_engine
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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
…
BUILD.gn
Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
2016-05-04 09:37:50 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +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
…
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
transport.h
…
typedefs.h
…
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ )
2016-05-04 08:54:39 +00:00
webrtc.gyp
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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.