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webrtc / src.git / ebe36efad70d70848964a7d8349e71809f69093b
commitebe36efad70d70848964a7d8349e71809f69093b[log] [tgz]
authorTaylor Brandstetter <deadbeef@webrtc.org>Tue Oct 10 00:16:54 2017
committerCommit Bot <commit-bot@chromium.org>Tue Oct 10 17:07:28 2017
tree1666dfc2843094840c173bcc53485d7bbe67e23d
parent933d8b07ea8ec5d1b2a04de277a656a6713e9d4b [diff]
Update Java MediaStream when native stream's set of tracks changes.

This will handle the scenario where, for example, the initial
offer/answer only negotiates audio, and video is added later (to the
same stream). Previously, there was absolutely no way to get a handle to
the new track without hacking the SDP. Now, the stream will be updated
after setRemoteDescription finishes.

Bug: webrtc:5677
Change-Id: Iea31bb7744da6b82afdaf44c8f74d721298a9474
Reviewed-on: https://webrtc-review.googlesource.com/6261
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20228}
6 files changed
tree: 1666dfc2843094840c173bcc53485d7bbe67e23d
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  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. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info