Add lock to guard rtp packet sequencer.

With deferred packet sequencing, the PacketSequencer instance is called
directly from the RtpRtcp module while before it was called from within
the RTPSender while holding a lock.

Since sequence number assignment happens on the same thread as actual
packet sending, though thought was that locking was no longer needed.
Unfortunately, SetRtpState()/GetRtpState() also exists - and while they
should only be called on creating/destruction there is a possible race
where a delayed packet from the pacer accesses the sequencer while
GetRtpState() is being called.

For now, this CL just adds a lock to guard sequencer. Follow-ups will
make sure get/set state is never called while module is attached to
the packet router. After that, the lock can be removed again.

Bug: webrtc:11340, webrtc:12470
Change-Id: I123c762fb4afd20b3a6bd03b86234eb9ec34a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228430
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34723}
2 files changed
tree: 04813639667cbd021ab2203f8db4fe069cfdf979
  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. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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