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pbos@webrtc.org dd8f6f3d48 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. 
_t names are reserved in POSIX.

BUG=162
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:18:42 +00:00
..
base
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
2014-12-18 04:45:05 +00:00
build
Move isolate path into webrtc/build/android/test_runner.py
2014-12-11 11:59:46 +00:00
common_audio
Make safe_conversions suitable for rtc_base_approved.
2014-12-17 22:56:09 +00:00
common_video
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
2014-12-18 02:28:25 +00:00
modules
Rename rtpDumpPktHdr_t to RtpDumpPacketHeader.
2014-12-18 09:18:42 +00:00
overrides
Remove or rename typedefs with _t prefixes.
2014-12-17 13:43:55 +00:00
p2p
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
2014-12-18 04:45:05 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Add video send bitrates to histogram stats:
2014-12-09 09:47:53 +00:00
test
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
tools
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +00:00
video
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
video_engine
Change aggregated fraction loss to be calculated from the cumulative loss and extended sequence number diff between the current and the last report block of two get stats calls.
2014-12-17 10:27:57 +00:00
voice_engine
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
2014-12-15 22:09:40 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Wire up RTT statistics to webrtc::Call.
2014-12-11 13:26:09 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add field to counters for when first rtp/rtcp packet is sent/received.
2014-12-16 12:03:11 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Add AGC manager tests.
2014-12-16 14:48:47 +00:00
webrtc.gyp
Merge in AGC manager and AGC tools.
2014-12-15 16:33:16 +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.