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}
diff --git a/webrtc/webrtc_examples.gyp b/webrtc/webrtc_examples.gyp
index 44b2ca3..fd39b03 100755
--- a/webrtc/webrtc_examples.gyp
+++ b/webrtc/webrtc_examples.gyp
@@ -14,7 +14,7 @@
'target_name': 'relayserver',
'type': 'executable',
'dependencies': [
- '../talk/libjingle.gyp:libjingle',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
],
'sources': [
@@ -25,7 +25,7 @@
'target_name': 'stunserver',
'type': 'executable',
'dependencies': [
- '../talk/libjingle.gyp:libjingle',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
],
'sources': [
@@ -36,7 +36,7 @@
'target_name': 'turnserver',
'type': 'executable',
'dependencies': [
- '../talk/libjingle.gyp:libjingle',
+ '<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
],
'sources': [
@@ -56,8 +56,8 @@
'examples/peerconnection/server/utils.h',
],
'dependencies': [
+ '<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/common.gyp:webrtc_common',
- '../talk/libjingle.gyp:libjingle',
],
# TODO(ronghuawu): crbug.com/167187 fix size_t to int truncations.
'msvs_disabled_warnings': [ 4309, ],
@@ -80,7 +80,6 @@
'dependencies': [
'../talk/libjingle.gyp:libjingle_peerconnection',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
- '<@(libjingle_tests_additional_deps)',
],
'conditions': [
['build_json==1', {