Forward fixing WebRTC to compile with Android NDK r16.

Starting from Chromium Roll [1], WebRTC should start to use NDK r16
for Android builds. The roll cannot be completed because of three
compilation errors:

../../sdk/android/src/jni/pc/androidnetworkmonitor.cc:15:9: error: 'RTLD_NOLOAD' macro redefined [-Werror,-Wmacro-redefined]
        ^
../../third_party/android_tools/ndk/sysroot/usr/include/dlfcn.h:62:9: note: previous definition is here

../../modules/audio_device/android/audio_record_jni.cc:251:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
  ALOGD("direct buffer capacity: %lld", capacity);

../../modules/audio_device/android/audio_track_jni.cc:229:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
  ALOGD("direct buffer capacity: %lld", capacity);

This CL forward fixes these errors in order to fix the Chromium Roll
into WebRTC.

[1] - https://webrtc-review.googlesource.com/c/src/+/37540

Bug: webrtc:8710
Change-Id: I5bc64e73919eee7c9e965a442a386b5e1897b56a
Reviewed-on: https://webrtc-review.googlesource.com/37640
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21510}
3 files changed
tree: 7a4dae9d6d1079a4d672eeda1052079d5cdfd1b9
  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. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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