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johan 2d8d23eec2 RFC 3984 sprop-parameter-sets SDP unit test 
BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2023913002
Cr-Commit-Position: refs/heads/master@{#13026}
2016-06-03 08:22:47 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
RFC 3984 sprop-parameter-sets SDP unit test
2016-06-03 08:22:47 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Disabled two flaky subtests of SignalThreadTest.
2016-06-02 15:47:12 +00:00
build
Fix building iOS SDK static libraries.
2016-06-02 19:22:24 +00:00
call
Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )
2016-06-03 06:24:58 +00:00
common_audio
Add clz functions (Count number of Leading Zero bits), 32-and 64-bit variants
2016-06-02 11:02:22 +00:00
common_video
Add H264 bitstream rewriting to limit frame reordering marker in header
2016-06-02 09:43:38 +00:00
examples
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
2016-05-31 20:02:30 +00:00
libjingle
Remove test targets of type none.
2016-06-02 09:26:10 +00:00
media
Combining SetVideoSend and SetSource into one method.
2016-06-02 23:23:47 +00:00
modules
Reduce logging frequency in bwe simulations.
2016-06-03 07:20:40 +00:00
p2p
Remove test targets of type none.
2016-06-02 09:26:10 +00:00
pc
Combining SetVideoSend and SetSource into one method.
2016-06-02 23:23:47 +00:00
sdk
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
system_wrappers
GN: Exclude tools and tests in Chromium builds.
2016-06-02 11:29:13 +00:00
test
AudioDecoderIsacT: Require caller to always specify sample rate
2016-06-02 09:59:02 +00:00
tools
GN: Exclude tools and tests in Chromium builds.
2016-06-02 11:29:13 +00:00
video
Always send RED headers if configured.
2016-06-03 07:16:54 +00:00
voice_engine
AudioDecoderIsacT: Require caller to always specify sample rate
2016-06-02 09:59:02 +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/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #5 id:80001 of https://codereview.webrtc.org/1979933002/ )
2016-06-01 11:45:13 +00:00
call.h
…
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
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
…
config.cc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
config.h
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
…
video_encoder.h
…
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +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
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Remove test targets of type none.
2016-06-02 09:26:10 +00:00
webrtc.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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.