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mflodman@webrtc.org 1b32bbe0a7 Removing private and unused method in RTPReceiver. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42269004

Cr-Commit-Position: refs/heads/master@{#8650}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8650 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 11:54:45 +00:00
..
base
Fix build configuration bug with debug builds.
2015-03-08 07:39:22 +00:00
build
Fix build configuration bug with debug builds.
2015-03-08 07:39:22 +00:00
common_audio
Enabling common_audio building with NEON on ARM64
2015-03-05 19:14:21 +00:00
common_video
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
2015-03-06 12:38:22 +00:00
examples
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
libjingle
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
modules
Removing private and unused method in RTPReceiver.
2015-03-09 11:54:45 +00:00
overrides
Re-landing perf improvement for libjingle logging after reverting the general change.
2015-03-07 12:18:14 +00:00
p2p
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
test
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
2015-03-06 12:38:22 +00:00
tools
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
2015-03-06 12:38:22 +00:00
video_engine
Revert "Workaround Mac align bug for observer_ and crit_."
2015-03-09 10:43:44 +00:00
voice_engine
Remove WebRtcACMEncodingType
2015-03-06 07:51:21 +00:00
.gitignore
…
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Remove DCHECK from common_types.cc
2015-03-04 20:10:38 +00:00
common_types.h
Add CVO support to RTP sender side.
2015-03-04 22:55:43 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
…
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
…
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Remove webrtc/system_wrappers/interface/scoped_ptr.h
2015-03-03 10:17:49 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
…
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
…
video_frame.h
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
2015-03-06 12:38:22 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
2015-03-06 12:38:22 +00:00
webrtc_examples.gyp
Improve cleaning for Android demo applications
2015-02-28 11:18:20 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.