2015-01-20 21:36:13 +00:00
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/*
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2016-02-07 20:46:45 -08:00
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* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
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2015-01-20 21:36:13 +00:00
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*
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2016-02-07 20:46:45 -08:00
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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2015-01-20 21:36:13 +00:00
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*/
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2013-07-10 00:45:36 +00:00
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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}
2016-02-04 23:52:28 -08:00
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#include "webrtc/media/base/mediaengine.h"
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2013-07-10 00:45:36 +00:00
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2014-06-20 14:58:56 +00:00
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#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
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2014-06-20 00:26:50 +00:00
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2014-08-19 14:56:59 +00:00
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#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)
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2016-02-12 06:39:40 +01:00
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#include "webrtc/media/engine/webrtcmediaengine.h"
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2014-08-19 14:56:59 +00:00
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#endif // HAVE_WEBRTC_VOICE && HAVE_WEBRTC_VIDEO
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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2014-06-20 00:26:50 +00:00
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2014-06-20 14:58:56 +00:00
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MediaEngineFactory::MediaEngineCreateFunction
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MediaEngineFactory::create_function_ = NULL;
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2014-08-19 14:56:59 +00:00
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2014-03-03 18:30:11 +00:00
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MediaEngineFactory::MediaEngineCreateFunction
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MediaEngineFactory::SetCreateFunction(MediaEngineCreateFunction function) {
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MediaEngineCreateFunction old_function = create_function_;
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create_function_ = function;
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return old_function;
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2014-08-19 14:56:59 +00:00
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}
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2014-03-03 18:30:11 +00:00
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2013-07-10 00:45:36 +00:00
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}; // namespace cricket
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#endif // DISABLE_MEDIA_ENGINE_FACTORY
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2016-03-16 19:07:43 -07:00
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namespace cricket {
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webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
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webrtc::RtpParameters parameters;
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webrtc::RtpEncodingParameters encoding;
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parameters.encodings.push_back(encoding);
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return parameters;
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}
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2016-03-29 17:47:19 +02:00
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}; // namespace cricket
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