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tfarina 90d67ddc1d Remove two more deprecated methods from SocketAddress API. 
This patch removes IPToString and IPToSensitiveString static helper
methods, since there are class methods that replace them already, and
they aren't used by anyone anymore.

BUG=None
R=pthacher@webrtc.org

Review URL: https://codereview.webrtc.org/1408873005

Cr-Commit-Position: refs/heads/master@{#10391}
2015-10-23 18:22:06 +00:00
..
audio
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
base
Remove two more deprecated methods from SocketAddress API.
2015-10-23 18:22:06 +00:00
build
Suppress libyuv::TestCpuFlag races.
2015-10-20 11:37:41 +00:00
call
Set send times in send time history via OnSentPacket.
2015-10-23 09:05:43 +00:00
common_audio
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
common_video
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
examples
Android: Replace EGL14 with EGL10
2015-10-22 23:52:45 +00:00
libjingle
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
modules
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
p2p
Separate StunProber::Start into Prepare and Run so we could create multiple of them and send out STUN pings at regular interval.
2015-10-23 02:16:02 +00:00
sound
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
system_wrappers
clean up field_trial_default target, to be used by remoting_perftests.
2015-10-23 02:33:22 +00:00
test
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
tools
Fix minor GYP error in webrtc/tools/internal_tools.gyp
2015-10-07 12:51:40 +00:00
video
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_engine
Rename ChannelGroup to CongestionController and move to webrtc/call/.
2015-10-21 13:52:33 +00:00
voice_engine
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
.gitignore
…
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
BUILD.gn
Update build files to use webrtc_overrides in Chromium instead of overrides.
2015-09-30 10:49:11 +00:00
call.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
codereview.settings
Update lower-level codereview.settings files.
2015-10-06 21:19:34 +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
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
common.gyp
…
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +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
Revert of Adding support for simulcast and spatial layers into VideoQualityTest (patchset #10 id:180001 of https://codereview.webrtc.org/1353263005/ )
2015-10-12 13:33:24 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
libjingle_examples.gyp
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow encoders to fall back dynamically to software.
2015-10-22 06:54:57 +00:00
video_engine_tests.isolate
…
video_frame.h
Remove VideoFrameType aliases for FrameType.
2015-10-23 13:58:27 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
2015-10-16 21:35:11 +00:00
webrtc.gyp
Remove references to libpeerconnection.
2015-10-08 12:41:05 +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.