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webrtc / src / ceb71cd557ad6c1c128132a55df9f781bca1e1cd
commitceb71cd557ad6c1c128132a55df9f781bca1e1cd[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Mon Sep 12 09:40:04 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Sep 12 12:43:17 2022
tree9caefc0ffffa8706b2fa43d9b9b4d9713b536784
parenta22c2a0c581cbe3f612f7a7d9fb9840186cc1e06 [diff]
Switch encoder on any critical frame encode error (returncode < 0)

Before this change encoder switch was triggered only if encode() returns WEBRTC_VIDEO_CODEC_ENCODER_FAILURE. Android HW encoder wrapper never return this error code. It returns WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE [1, 2] or WEBRTC_VIDEO_CODEC_ERROR [3].

Change value of WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT from -14 to 5 to avoid it to be interpreted as a critical error.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=324

[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=335

[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/video_encoder_wrapper.cc;drc=c05a1be5b49f5c03b6955b05bcbf47609e1b0381;l=331

Bug: b/243402636
Change-Id: Iaf0129f3f9d71c07bb06804fe1f92a1f84f6da26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274402
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38060}
3 files changed
tree: 9caefc0ffffa8706b2fa43d9b9b4d9713b536784
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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