Chih-Hung Hsieh 675d4373f8 WIP: Changes after merge commit 'cb3f9bd'
Changes after "git merge cb3f9bd"

* git mv old Android.mk from src/ to webrtc/
* Remove old unused files in src/*.
* Modify webrtc/.gitignore to keep *.mk files.
* Copy old files from master, lost in auto-merge.
      src/modules/audio_processing/test/unit_test.cc
      src/modules/audio_coding/codecs/isac/fix/test/{Android.mk,kenny.c}
  to webrtc, but most of the old test code do not compile with new
  webrtc API and are commented out.
* Move src/modules/audio_processing/test/android/apmtest/jni/*.mk to
  webrtc/... but the Android.mk files does not work.
  Commented out its build target.
* Changes to Android.mk files:
  * Change references of src/ to webrtc/.
  * Fix include path
  * Fix source file list, remove old non-existing files,
    add new source files to resolve link errors.
  * Add new Android.mk files to build some new static libraries
    to link into current Android webrtc .so files.
  * Remove unnecessary LOCAL_SHARED_LIBRARIES in Android.mk files
    that build static libraries.
  * Remove old unnecessary clang workarounds like
     -Wno-tautological-pointer-compare
     -no-integrated-as
* Fix include path of debug.pb.h in some source files.
* Add -DWEBRTC_POSIX in android-webrtc.mk
* Manually merge Android specific changes in
     src/typedefs.h to webrtc/typedefs.h
* Fix trivial syntax error in scoped_ptr.h, calling static_assert.
* Use -std=c++0x in webrtc/system_wrappers/source/Android.mk
* #undef getchaar in spreadsort.hpp
* Verified and not to carry old Android hacks from src/... to webrtc/...
      src/system_wrappers/source/android/cpu-features.c
      src/modules/interface/module.h
      src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
      src/system_wrappers/source/trace_posix.cc
      src/typedefs.h

More pathes from Alex Luebs:
* Use new unit test kenny.cc.
  Delete old kenny.cc.
  Comment out unessential code in kenny.cc to fix link error for now.
* Replace old unit test files with new ones in
  webrtc/modules/audio_processing/Android.mk.
  Delete old audio_processing/test/unit_test.cc.
* Fix compilation errors in
  webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Change-Id: I7bbf776eeb9dcfa21a82dd1f2dec378235cbbc3e
2015-11-25 11:43:05 -08:00
2015-09-22 06:09:26 +00:00
2014-06-17 08:54:03 +00:00
2012-02-17 10:26:11 -08:00
2015-09-11 09:04:09 +00:00
2012-08-21 16:37:02 +01:00
2015-09-25 13:16:38 +00:00

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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

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