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Sebastian Jansson 800e121dca Adds support to change transport routes in Scenario tests.
This CL makes it possible to change transport routes while running
a scenario based test.

To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.

Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
2018-10-22 11:14:37 +00:00
api Audio codecs API: Remove some weasel words in the docs 2018-10-22 08:52:15 +00:00
audio Media engine and channel support for per-channel dscp values, specified by RtpParameter 2018-10-19 21:47:55 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Media engine and channel support for per-channel dscp values, specified by RtpParameter 2018-10-19 21:47:55 +00:00
common_audio Extract functionality of test_main into separate library. 2018-10-15 14:13:06 +00:00
common_video Delete dead code in webrtc_libyuv.cc 2018-10-22 08:41:50 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Reland "Prefix flag macros with WEBRTC_." 2018-10-19 15:06:43 +00:00
infra Remove ios32_sim_ios9_dbg from CQ. 2018-10-15 06:59:19 +00:00
logging Add blob-encoding support for RTC event logs 2018-10-19 17:30:02 +00:00
media Media engine and channel support for per-channel dscp values, specified by RtpParameter 2018-10-19 21:47:55 +00:00
modules Adds field trial to do safer reset on route change. 2018-10-22 10:46:49 +00:00
p2p Replace _stricmp with absl::EqualsIgnoreCase 2018-10-19 14:17:31 +00:00
pc Media engine and channel support for per-channel dscp values, specified by RtpParameter 2018-10-19 21:47:55 +00:00
resources Drop legacy AEC metrics interface from ApmTest.Process 2018-09-14 08:16:43 +00:00
rtc_base Reland "Prefix flag macros with WEBRTC_." 2018-10-19 15:06:43 +00:00
rtc_tools Reland "Prefix flag macros with WEBRTC_." 2018-10-19 15:06:43 +00:00
sdk Fix a null reference bug in NetworkMonitorAutoDetect.getNetworkState. 2018-10-19 21:18:35 +00:00
stats Move rtc_json code from API dir, enable unit test, unmark testonly 2018-10-02 15:21:26 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reland "Make sure Chromium will pick the correct field_trial/metric impl." 2018-10-22 07:25:32 +00:00
test Adds support to change transport routes in Scenario tests. 2018-10-22 11:14:37 +00:00
tools_webrtc Remove expat from generate_licenses.py. 2018-10-18 09:02:54 +00:00
video Reland "Prefix flag macros with WEBRTC_." 2018-10-19 15:06:43 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore format commit. 2018-06-20 09:26:44 +00:00
.gitignore Reland "Compile frame analyzer for the host machine on perf tests." 2018-09-18 09:51:19 +00:00
.gn Re-enable gtest absl pretty printers. 2018-08-13 13:54:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Replace _stricmp with absl::EqualsIgnoreCase 2018-10-19 14:17:31 +00:00
AUTHORS Generate module.modulemap file when building Mac Framework 2018-10-08 12:32:45 +00:00
BUILD.gn Replace _stricmp with absl::EqualsIgnoreCase 2018-10-19 14:17:31 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.h Replace _stricmp with absl::EqualsIgnoreCase 2018-10-19 14:17:31 +00:00
DEPS Replace _stricmp with absl::EqualsIgnoreCase 2018-10-19 14:17:31 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Add documentation about field_trial/metrics custom impl. 2018-09-18 11:27:59 +00:00
OWNERS Add a list of allowed and disallowed Abseil things 2018-08-27 08:37:20 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Allow the autoroller service account to commit 2018-10-04 09:25:03 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Add a list of allowed and disallowed Abseil things 2018-08-27 08:37:20 +00:00
WATCHLISTS Adds srte to WATCHLISTS. 2018-10-11 11:52:28 +00:00
webrtc.gni Add rtc_enable_symbol_export to incrementally create a WebRTC component. 2018-10-10 11:04:34 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info