Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.

This CL adds the possibility to collect the following Video BWE stats:
- available_send_bandwidth
- transmission_bitrate
- retransmission_bitrate
- actual_encode_bitrate
- target_encode_bitrate

Example of the output:

RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond

Bug: webrtc:10138
Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27760}
4 files changed
tree: 5f210cfeb6213d9fe51865b5672ce5030ec10735
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  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. 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