Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone."

This reverts commit 05d43c6f7fe497fed0f2c8714e2042dd07a86df2.

Reason for revert: It breaks some Chromium trybots:
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_asan_rel_ng/206387
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_tsan_rel_ng/207737
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/win10_chromium_x64_rel_ng/202283

Original change's description:
> Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
> 
> PeerConnection::Close() is, per-spec, a blocking operation.
> Unfortunately, PeerConnection is implemented to own resources used by
> the network thread, and Close() - on the signaling thread - destroys
> these resources. As such, tasks run in parallel like getStats() get into
> race conditions with Close() unless synchronized. The mechanism in-place
> is RTCStatsCollector::WaitForPendingRequest(), it waits until the
> network thread is done with the in-parallel stats request.
> 
> Prior to this CL, this was implemented by performing
> rtc::Thread::ProcessMessages() in a loop until the network thread had
> posted a task on the signaling thread to say that it was done which
> would then get processed by ProcessMessages(). In WebRTC this works, and
> the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
> 
> But because Chromium's thread wrapper does no support
> ProcessMessages(), calling getStats() followed by close() in Chrome
> resulted in waiting forever (https://crbug.com/850907).
> 
> In this CL, the process messages loop is removed. Instead, the shared
> resources are guarded by an rtc::Event. WaitForPendingRequest() still
> blocks the signaling thread, but only while shared resources are in use
> by the network thread. After this CL, calling WaitForPendingRequest() no
> longer has any unexpected side-effects since it no longer processes
> other messages that might have been posted on the thread.
> 
> The resource ownership and threading model of WebRTC deserves to be
> revisited, but this fixes a common Chromium crash without redesigning
> PeerConnection, in a way that does not cause more blocking than what
> the other PeerConnection methods are already doing.
> 
> Note: An alternative to using rtc::Event is to use resource locks and
> to not perform the stats collection on the network thread if the
> request was cancelled before the start of processing, but this has very
> little benefit in terms of performance: once the network thread starts
> collecting the stats, it would use the lock until collection is
> completed, blocking the signaling thread trying to acquire that lock
> anyway. This defeats the purpose and is a riskier change, since
> cancelling partial collection in this inherently racy edge-case would
> have observable differences from the returned stats, which may cause
> more regressions.
> 
> Bug: chromium:850907
> Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
> Reviewed-on: https://webrtc-review.googlesource.com/c/121567
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26707}

TBR=steveanton@webrtc.org,hbos@webrtc.org

Change-Id: Icd82cdd5bd086a90999f7fd5f8616e1f2d2153bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:850907
Reviewed-on: https://webrtc-review.googlesource.com/c/123225
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26721}
4 files changed
tree: 22e9c68f8549605f1ddb0a3f2a9848d9eac2c465
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. logging/
  10. media/
  11. modules/
  12. p2p/
  13. pc/
  14. resources/
  15. rtc_base/
  16. rtc_tools/
  17. sdk/
  18. stats/
  19. style-guide/
  20. system_wrappers/
  21. test/
  22. tools_webrtc/
  23. video/
  24. .clang-format
  25. .git-blame-ignore-revs
  26. .gitignore
  27. .gn
  28. .vpython
  29. abseil-in-webrtc.md
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.h
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. 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 http://www.webrtc.org/native-code/development 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