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Taylor Brandstetter 55dd70842c Support RtpEncodingParameters::active in voice engine. 
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1943073003 .

Cr-Commit-Position: refs/heads/master@{#12615}
2016-05-03 20:50:24 +00:00
..
api
Android EGL: Synchronize calls to eglCreateContext
2016-05-03 08:24:44 +00:00
audio
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
base
Rename rtc::Time64 --> rtc::TimeMillis.
2016-05-02 15:19:00 +00:00
build
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
call
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common_audio
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
common_video
This cl deletes the class webrtc::VideoRendererCallback.
2016-05-03 08:09:17 +00:00
examples
VideoCapturerAndroid: Remove deprecated create function with egl context argument
2016-05-02 07:43:32 +00:00
libjingle
Replace scoped_ptr with unique_ptr in webrtc/libjingle/
2016-04-30 09:40:26 +00:00
media
Support RtpEncodingParameters::active in voice engine.
2016-05-03 20:50:24 +00:00
modules
Reducing neteq sync buffer size.
2016-05-03 20:32:13 +00:00
p2p
Do not stop a session unless the candidate of a writable connection belongs to the
2016-05-03 00:28:42 +00:00
pc
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
sdk
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
test
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Re-enable Vp9FlexModeRefCount
2016-05-03 12:45:48 +00:00
voice_engine
Move ADM Create() method to public interface.
2016-05-03 19:51:31 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Enable -Winconsistent-missing-override flag.
2016-04-29 13:09:23 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
…
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
…
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
…
config.h
…
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
webrtc.gyp
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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.