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pbos@webrtc.org e02d47515f Set up receiver RTX config using a std::map. 
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-20 14:43:55 +00:00
..
build
Exclude protoc objects from merge_libs.py.
2014-01-16 00:31:57 +00:00
common_audio
Remove the requirement to call set_sample_rate_hz and friends.
2014-01-07 17:45:09 +00:00
common_video
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
examples
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
2014-01-17 23:12:51 +00:00
modules
Add configuration and test for extended RTCP reference time reports to new video api.
2014-01-20 08:34:49 +00:00
system_wrappers
Removes the remaining uses of the list wrapper class and the list wrapper class.
2014-01-13 22:41:34 +00:00
test
Removes usage of ListWrapper from several files.
2014-01-13 15:21:30 +00:00
tools
Now printing less output from compare_videos.py.
2014-01-07 17:59:30 +00:00
video
Add configuration and test for extended RTCP reference time reports to new video api.
2014-01-20 08:34:49 +00:00
video_engine
Add configuration and test for extended RTCP reference time reports to new video api.
2014-01-20 08:34:49 +00:00
voice_engine
Temporarily disabling some more audio processing tests.
2014-01-14 08:57:22 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Remove outdated DestroyVideoSendStream comment.
2014-01-07 12:27:22 +00:00
common_types.h
Remove empty VideoCodecGeneric struct.
2014-01-14 12:55:59 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Set up receiver RTX config using a std::map.
2014-01-20 14:43:55 +00:00
engine_configurations.h
Adds a new voice engine warning for the typing noise off state.
2013-09-17 18:09:20 +00:00
experiments.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
frame_callback.h
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +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
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
2013-12-10 19:20:46 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Set up receiver RTX config using a std::map.
2014-01-20 14:43:55 +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 statistics in video send stream of new video engine api
2014-01-07 09:54:34 +00:00
webrtc_examples.gyp
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
2014-01-17 23:12:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.