Reland "Allow unspecified max allocatable bitrate in VideoSendStreamImpl"

This is a reland of commit e844bd819c4663de5422be9c2cdb25204b03b41b

The changes after Patchset 1 are intended to fix a corner case where a
video send stream config is used where layers (simulcast or svc) are
configured and active but have undefined max bitrate. In those cases the
max bitrate will be inherited from the global max bitrate.

Original change's description:
> Allow unspecified max allocatable bitrate in VideoSendStreamImpl
>
> If we have no video the can be enabled, based on the VideoEncoderConfig
> (e.g. the max bitrate is <= 0 or all simulcast streams are
> `active=false`) - let the max bitrate be reported as 0 for the sake of
> bandwidth allocation (MediaStreamAllocationConfig).
>
> Previous to this CL, the above conditions would result in an arbitrary
> value of 10Mbps to use as the max allocation limit. When creating a peer
> connection with audio and video configured but the video disabled/muted,
> that would result in the bandwidth estimator trying to send packet
> trains to probe up to 10Mbps, potentially impacting the network and
> disturbing audio even though we have no need for such as high BWE.
>
> Bug: webrtc:494350649
> Change-Id: Id6353fd93c6170610a790c084d75f3b3c5c3ee97
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/449100
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47220}

Bug: webrtc:494350649
Change-Id: I59e415bbd121b8661dc5b53e67a2788a335539a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459680
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47245}
6 files changed
tree: 7e606a4de9f0a5a6ccc44254eaa27366c06189a5
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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