kjellander a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00
..
2016-02-05 07:52:35 +00:00

This directory holds a Java implementation of the webrtc::PeerConnection API, as
well as the JNI glue C++ code that lets the Java implementation reuse the C++
implementation of the same API.

To build the Java API and related tests, build with 
OS=linux or OS=android and include
build_with_libjingle=1 build_with_chromium=0
in $GYP_DEFINES.

To use the Java API, start by looking at the public interface of
org.webrtc.PeerConnection{,Factory} and the org.webrtc.PeerConnectionTest.

To understand the implementation of the API, see the native code in jni/.

An example command-line to build & run the unittest:
cd path/to/trunk
GYP_DEFINES="build_with_libjingle=1 build_with_chromium=0 java_home=path/to/JDK" gclient runhooks && \
    ninja -C out/Debug libjingle_peerconnection_java_unittest && \
    ./out/Debug/libjingle_peerconnection_java_unittest
(where path/to/JDK should contain include/jni.h)

During development it can be helpful to run the JVM with the -Xcheck:jni flag.