Reduce thread hops in StatsCollector and fix incorrect variable access.

StatsCollector::ExtractSessionInfo was run fully on the signaling thread
and several calls were being made to methods that need to run on the
network thread.

Additionally, BaseChannel::transport_name() was being read directly
on the signaling thread (needs to be read on the network thread).
So with shifting the work that needs to happen on the network thread
over to that thread, we now also grab the transport name there and
use the name with the work that still needs to happen on the signaling
thread.

These changes allow us to remove Invoke<>() calls to the network thread from
callback functions implemented in PeerConnection:
* GetPooledCandidateStats
* GetTransportNamesByMid
* GetTransportStatsByNames
* Also adding a correctness thread check to:
  * GetLocalCertificate
  * GetRemoteSSLCertChain

Because PeerConnection now has a way of knowing when things are
or have been uninitialized on the network thread, all of these
functions can exit early without doing throw away work.

Additionally removing thread hops that aren't needed anymore from
JsepTransportController.

Using the RTC_LOG_THREAD_BLOCK_COUNT() macro in GetStats, the number
of Invokes (when >1), goes down by 3. Typically from 8->5, 7->4, 6->3.

Bug: webrtc:11687
Change-Id: I06ab25eab301e192e99076d7891444bcb61b491f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214135
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33656}
4 files changed
tree: d6c5e890f844ab15c389a6d1f1a4e5c0991e2147
  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. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  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. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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