Add a field trial to control datagram transport use.

First, the existing configuration parameter (use_datagram_transport) is
now optional.

The new field trial has two flag values:
 1. Whether to enable the datagram transport (enabled)
 2. Whether to use the datagram transport by default (default_value)

The first is a kill-switch.  It disables the datagram transport, even
for applications which inject a datagram transport factory and specify
use_datagram_transport = true.  This allows applications which hard-code
a datagram transport to switch it off via field trials.

This flag defaults to true, to avoid breaking downstream projects which
already inject and configure a datagram transport.  It may be changed to
false after updating downstream to set this field trial flag to true
when required.

The second provides a default value to be used in case the
aforementioned use_datagram_transport parameter is unset.  Applications
which explicitly set use_datagram_transport will use that value.
Applications which do not explicitly specify whether or not to use the
datagram transport will use it (or not) according to the default_value
flag.

One goal of this flag is to simplify rollout in applications which
already set field trials based on configuration, but require code
changes for new RTCConfiguration parameters.  A second goal is to
provide platforms with a knob to control whether datagram transport is
"opt-in" or "opt-out".

This flag defaults to false, to prevent downstream projects from
unintentionally enabling the datagram tranpsort.

Bug: webrtc:9719
Change-Id: I521a5fa61c992e76e5081118678a1812a261d672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28435}
3 files changed
tree: 7a7e3f37d3287ed6232ede45e5c76b2c6efb75f2
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. whitespace.txt
README.md

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