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webrtc / src / 7517fb639b69753ce1f2a5c1c07b694342cece11
commit7517fb639b69753ce1f2a5c1c07b694342cece11[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Tue Jun 14 10:40:19 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Jun 16 15:55:09 2022
treede55db5834c496dcda075ac7d16b1de841763f93
parentef3137a9287429b64706ce2fc85d5982f6d8ca9a [diff]
Switch to getInput/OutputBuffer

Use getInput/OutputBuffer(index) instead of getInput/OutputBuffers() in
Android MediaCodec video encoder and decoder wrappers.

getInput/OutputBuffers(index) are available from SDK 21 which is the minimum required version in WebRTC: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/AndroidManifest.xml

Bug: b/234879577
Change-Id: I79fd234b104420ae3544229e8c62d7db2344cd01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265804
Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37241}
7 files changed
tree: de55db5834c496dcda075ac7d16b1de841763f93
  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