Reland "Add trace of enqueued and sent RTP packets"

This reverts commit 45bb717a2866c2d836b5332a24af0d09f2b30714.

Reason for revert: Use #if RTC_TRACE_EVENTS_ENABLED to avoid unused variable.

Original change's description:
> Revert "Add trace of enqueued and sent RTP packets"
> 
> This reverts commit 45b9192ad981dcdc12ad4aef087fff2195bd030c.
> 
> Reason for revert: When tracing is disabled, this results in a clang warning (unused variable), which results in a build error since Werror is enabled by default.
> 
> Original change's description:
> > Add trace of enqueued and sent RTP packets
> > 
> > This is useful in debugging the latency from a packet
> > is enqueued until it's sent.
> > 
> > Bug: webrtc:11617
> > Change-Id: Ic2f194334a2e178de221df3a0838481035bb3505
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176231
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31381}
> 
> TBR=sprang@webrtc.org,kron@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:11617
> Change-Id: I854c17e587c624691a0e5e3ec9fd38c2607eda84
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176380
> Commit-Queue: Casey Fischer <caseyfischer@google.com>
> Reviewed-by: Adam Nathan <adamnathan@google.com>
> Cr-Commit-Position: refs/heads/master@{#31399}

TBR=sprang@webrtc.org,yujo@chromium.org,adamnathan@google.com,kron@webrtc.org,caseyfischer@google.com

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11617
Change-Id: I9de7f7ed290481a51c161a693f5b2d5df7d2eae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176367
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31407}
3 files changed
tree: bb9981fbd63f73b17e0ffbba74d19d39cd249352
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.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