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webrtc / src / 326df690b2a3e1bbb5d1d7f466d86bbec16bad68
commit326df690b2a3e1bbb5d1d7f466d86bbec16bad68[log] [tgz]
authorJianjun Zhu <jianjun.zhu@intel.com>Mon Apr 15 01:11:25 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Apr 15 09:06:12 2024
tree9e54bca73780ac54bcc66cc8219658bc86843a30
parent1261863f87ff0f6c7910a039012d3469ee8ee2fb [diff]
Use H26xPacketBuffer for H.264 and H.265 packets.

This CL updates RtpVideoStreamReceiver2 to use H26xPacketBuffer for
H.264 and H.265 packets. H.264 specific fixes are moved to
H26xPacketBuffer as well.

H26xPacketBuffer is behind field trial WebRTC-Video-H26xPacketBuffer.

Bug: webrtc:13485
Change-Id: I1874c5a624b94c2d75ce607cf10c939619d7b5b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42062}
5 files changed
tree: 9e54bca73780ac54bcc66cc8219658bc86843a30
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  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. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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