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platform-external-webrtc/webrtc
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Henrik Boström faa78dc38d Removed old DtlsIdentityRequestObserver::RequestIdentity function signature 
since the new signature is used everywhere.

BUG=chromium:544902, webrtc:5092
R=mattdr@google.com, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1964663004 .

Cr-Commit-Position: refs/heads/master@{#12681}
2016-05-11 08:23:34 +00:00
..
api
Removed old DtlsIdentityRequestObserver::RequestIdentity function signature
2016-05-11 08:23:34 +00:00
audio
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
base
rtc::Optional<T>: Don't secretly contain a default-constructed T when empty
2016-05-09 13:06:10 +00:00
build
Revert of Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://cod… (patchset #1 id:1 of https://codereview.webrtc.org/1944273002/ )
2016-05-11 06:28:22 +00:00
call
Revert "Reland of Remove SendPacer from ViEEncoder
2016-05-09 11:57:18 +00:00
common_audio
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
common_video
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
examples
Add test annotation to PeerConnectionClientTest.testLoopbackVp9DecodeToTexture test.
2016-05-09 17:15:24 +00:00
libjingle
Change default timestamp to 64 bits in all webrtc directories.
2016-05-06 18:29:27 +00:00
media
Allow WebRTC to offer receiving capability for 120ms Opus packets.
2016-05-10 13:35:30 +00:00
modules
Reland of name OpenH264 frame-type conversion function. (patchset #1 id:1 of https://codereview.webrtc.org/1964913002/ )
2016-05-11 07:58:42 +00:00
p2p
Fix AllocationSequence to handle the case when TurnPort stops using shared socket.
2016-05-10 17:20:54 +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
Delete all use of tick_util.h.
2016-05-10 14:31:58 +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
Delete all use of tick_util.h.
2016-05-10 14:31:58 +00:00
voice_engine
Delete all use of tick_util.h.
2016-05-10 14:31:58 +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
Revert of Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://cod… (patchset #1 id:1 of https://codereview.webrtc.org/1944273002/ )
2016-05-11 06:28:22 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
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
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
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.