Refactor WebRTC SDP tests and deprecate old APIs

This refactors the SDP unit tests to use `SessionDescriptionInterface`
instead of `JsepSessionDescription` directly.

The `Initialize()` method is being deprecated, as it's an optional
two-step initialization, which can lead to indeterminate state.
Previously the tests used this and called Initialize() multiple times on
the same object, which made the tests harder to reason about. The tests
have been updated to use the `CreateSessionDescription()` factory method
instead and other api/ methods.

The change also:
* Adds a `RTC_DCHECK` to verify that no pre-existing candidates exist before adding new ones.
* Deprecates the `JsepSessionDescription(SdpType type)` constructor.
* Adds a new `add()` method to `IceCandidateCollection` to move candidates from another collection.
* Adds checks to `Clone()` to flag when candidates were not being cloned correctly.
* Significantly (but necessarily) refactors the SDP unit tests for better readability and to remove duplicate code.

Bug: webrtc:442220720
Change-Id: If78b400b4f53195786f7d61f41998e7ea6912163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408100
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45589}
12 files changed
tree: f115d9d7b8c5f491de66de98928e79dced23bc26
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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