Revert "Batch SDP candidates and O(1) TransportInfo lookup"

This reverts commit b38573bb56e571405b318c761c23a5b247d24716.

Reason for revert: Broke the chromium roll

Failure Link: https://chromium-review.git.corp.google.com/c/chromium/src/+/8622522?tab=checks

Original change's description:
> Batch SDP candidates and O(1) TransportInfo lookup
>
> SdpDeserialize added candidates one at a time with AddCandidate(), which
> looks up the m= section, copies the candidate and recomputes the
> connection address on every call. SdpSerialize copied the candidates of
> each m= section and looked up its TransportInfo by name.
>
> Now the parser fills one IceCandidateCollection per m= section and
> passes them to SessionDescriptionInterface::Create(), which updates each
> section's connection address once. The serializer reads the candidate
> collections directly and tries the TransportInfo at the same index
> before falling back to a lookup by name.
>
> BM_Sdp* benchmarks (release build, median of 5 runs):
> - Deserialize: -2.9% (small), -4.6% (large)
> - Serialize: -3.7% (small), -12.2% (large)
>
> Bug: none
> Change-Id: I04eae8cab5870cf60708cea9668cc147807ba176
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/509664
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#49014}

Bug: none
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I361a5616f1d51402b270f8a3d180453b610293d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/510940
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#49024}
5 files changed
tree: 1a920f51f48b049ee93a330a7e631b07092b4373
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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