webrtc_m130/webrtc/media/base/mediaengine.cc

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/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
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
#include "webrtc/media/base/mediaengine.h"
#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)
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
#include "webrtc/media/webrtc/webrtcmediaengine.h"
#endif // HAVE_WEBRTC_VOICE && HAVE_WEBRTC_VIDEO
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif // HAVE_CONFIG
namespace cricket {
MediaEngineFactory::MediaEngineCreateFunction
MediaEngineFactory::create_function_ = NULL;
MediaEngineFactory::MediaEngineCreateFunction
MediaEngineFactory::SetCreateFunction(MediaEngineCreateFunction function) {
MediaEngineCreateFunction old_function = create_function_;
create_function_ = function;
return old_function;
}
}; // namespace cricket
#endif // DISABLE_MEDIA_ENGINE_FACTORY