Add support for time-varying constraints in DegradedCall.

The fake network configs are now specified using just two field trials:
WebRTC-FakeNetworkSendConfig and WebRTC-FakeNetworkReceiveConfig.
Both of them have the following parameters from
BuiltInNetworkBehaviorConfig:

* queue_length_packets  // Queue length in number of packets.
* queue_delay_ms  // Delay in addition to capacity induced delay.
* delay_standard_deviation_ms   // Standard deviation of the extra delay.
* link_capacity_kbps  // Link capacity in kbps.
* loss_percent  // Random packet loss.
* allow_reordering  // If packets are allowed to be reordered.
* avg_burst_loss_length  // The average length of a burst of lost packets.
* packet_overhead  // Additional bytes to add to packet size.
* codel_active_queue_management  // Enable CoDel active queue management.

Plus:
* duration  // For how long to use this config before progressing.

Example:
WebRTC-FakeNetworkSendConfig/queue_delay_ms:66|1,loss_percent:1|0,link_capacity_kbps:200|10000,queue_length_packets:10|100,duration:15s|45s/
This creates two configs:
1. For 15s, apply 66ms delay, 1% loss, 200kbps bandwidth, 10 packet queue size
2. For 45s, apply 1ms delay, 0% loss, 10Mbps bandwidth, 100 packets queue size
(then repeat)

Bug: webrtc:13655
Change-Id: I0524f572de480731df4d414724203772182c628b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251043
Reviewed-by: Stefan Holmer <holmer@google.com>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35952}
3 files changed
tree: 8ae06d15a3a39d5d5f39773d4b23a96905bd13f8
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .style.yapf
  31. .vpython
  32. .vpython3
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. WATCHLISTS
  53. webrtc.gni
  54. webrtc_lib_link_test.cc
  55. 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 here 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