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platform-external-webrtc/webrtc
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pbos@webrtc.org 008731868a Implement settable min/start/max bitrates in Call. 
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-25 14:03:34 +00:00
..
base
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
build
Build fix for MIPS Android Webview build.
2014-11-21 16:28:32 +00:00
common_audio
Build fix for MIPS Android Webview build.
2014-11-21 16:28:32 +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
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Add wav output capability to neteq_rtpplay
2014-11-24 14:50:53 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
Add jmi field for packets discarded due to network error
2014-11-17 19:42:14 +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
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
test
OWNERS: Remove tomasl@ and mallinath@
2014-11-24 10:05:05 +00:00
tools
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video
Implement settable min/start/max bitrates in Call.
2014-11-25 14:03:34 +00:00
video_engine
OWNERS: Remove tomasl@ and mallinath@
2014-11-24 10:05:05 +00:00
voice_engine
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
.gitignore
…
BUILD.gn
Change the static_library("webrtc") to a source set in the GN build.
2014-11-12 07:56:21 +00:00
call.h
Implement settable min/start/max bitrates in Call.
2014-11-25 14:03:34 +00:00
common_types.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +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
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +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
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
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
…
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
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +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
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +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
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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.