diff --git a/DEPS b/DEPS index 7f2225c5c8..1ec0583f66 100644 --- a/DEPS +++ b/DEPS @@ -23,6 +23,25 @@ deps_os = { }, } +# Define rules for which include paths are allowed in our source. +include_rules = [ + # Base is only used to build Android APK tests and may not be referenced by + # WebRTC production code. + '-base', + '-chromium', + '+external/webrtc/webrtc', # Android platform build. + '+gflags', + '+libyuv', + '+net', + '+talk', + '+testing', + '+third_party', + '+unicode', + '+usrsctplib', + '+webrtc', + '+vpx', +] + hooks = [ { # Check for legacy named top-level dir (named 'trunk'). diff --git a/talk/app/webrtc/DEPS b/talk/app/webrtc/DEPS deleted file mode 100644 index 69ecd0279e..0000000000 --- a/talk/app/webrtc/DEPS +++ /dev/null @@ -1,7 +0,0 @@ -include_rules = [ - "+talk/app/webrtc/objc", - "+webrtc/video_frame.h", - "+webrtc/api", - "+webrtc/base", - "+webrtc/media", -] diff --git a/webrtc/DEPS b/webrtc/DEPS deleted file mode 100644 index 292c99685b..0000000000 --- a/webrtc/DEPS +++ /dev/null @@ -1,47 +0,0 @@ -# Define rules for which include paths are allowed in our source. -include_rules = [ - # Base is only used to build Android APK tests and may not be referenced by - # WebRTC production code. - "-base", - "-chromium", - "+external/webrtc/webrtc", # Android platform build. - "+gflags", - "+libyuv", - "+testing", - "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. - # Individual headers that will be moved out of here, see webrtc: - "+webrtc/audio_receive_stream.h", - "+webrtc/audio_send_stream.h", - "+webrtc/audio_sink.h", - "+webrtc/audio_state.h", - "+webrtc/call.h", - "+webrtc/common.h", - "+webrtc/common_types.h", - "+webrtc/config.h", - "+webrtc/engine_configurations.h", - "+webrtc/frame_callback.h", - "+webrtc/stream.h", - "+webrtc/transport.h", - "+webrtc/typedefs.h", - "+webrtc/video_decoder.h", - "+webrtc/video_encoder.h", - "+webrtc/video_frame.h", - "+webrtc/video_receive_stream.h", - "+webrtc/video_renderer.h", - "+webrtc/video_send_stream.h", - - "+webrtc/base", - "+webrtc/modules/include", - "+webrtc/test", - "+webrtc/tools", -] - -# The below rules will be removed when webrtc: is fixed. -specific_include_rules = { - "audio_send_stream\.h": [ - "+webrtc/modules/audio_coding", - ], - "video_frame\.h": [ - "+webrtc/common_video", - ], -} diff --git a/webrtc/api/DEPS b/webrtc/api/DEPS deleted file mode 100644 index 956d4d95c8..0000000000 --- a/webrtc/api/DEPS +++ /dev/null @@ -1,23 +0,0 @@ -include_rules = [ - "+third_party/libyuv", - "+webrtc/base", - "+webrtc/common_video", - "+webrtc/media", - "+webrtc/p2p", - "+webrtc/pc", - "+webrtc/modules/audio_device", - "+webrtc/modules/rtp_rtcp", - "+webrtc/modules/video_coding", - "+webrtc/modules/video_render", - "+webrtc/system_wrappers", -] - -specific_include_rules = { - "androidtestinitializer\.cc": [ - "+base/android", # Allowed only for Android tests. - "+webrtc/voice_engine", - ], - "peerconnection_jni\.cc": [ - "+webrtc/voice_engine", - ] -} diff --git a/webrtc/audio/DEPS b/webrtc/audio/DEPS deleted file mode 100644 index 63711ab428..0000000000 --- a/webrtc/audio/DEPS +++ /dev/null @@ -1,10 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/voice_engine", - "+webrtc/modules/bitrate_controller", - "+webrtc/modules/congestion_controller", - "+webrtc/modules/pacing", - "+webrtc/modules/remote_bitrate_estimator", - "+webrtc/modules/rtp_rtcp", - "+webrtc/system_wrappers", -] diff --git a/webrtc/base/DEPS b/webrtc/base/DEPS deleted file mode 100644 index add7f38be5..0000000000 --- a/webrtc/base/DEPS +++ /dev/null @@ -1,11 +0,0 @@ -include_rules = [ - "+json", - "+third_party/jsoncpp", - "+webrtc/system_wrappers", -] - -specific_include_rules = { - "gunit_prod.h": [ - "+gtest", - ], -} diff --git a/webrtc/call/DEPS b/webrtc/call/DEPS deleted file mode 100644 index 0f9030853c..0000000000 --- a/webrtc/call/DEPS +++ /dev/null @@ -1,13 +0,0 @@ -include_rules = [ - "+webrtc/audio", - "+webrtc/base", - "+webrtc/modules/audio_coding", - "+webrtc/modules/bitrate_controller", - "+webrtc/modules/congestion_controller", - "+webrtc/modules/pacing", - "+webrtc/modules/rtp_rtcp", - "+webrtc/modules/utility", - "+webrtc/system_wrappers", - "+webrtc/voice_engine", - "+webrtc/video", -] diff --git a/webrtc/common_audio/DEPS b/webrtc/common_audio/DEPS deleted file mode 100644 index 2d2185750f..0000000000 --- a/webrtc/common_audio/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+third_party/openmax_dl", - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/common_audio/real_fourier_openmax.cc b/webrtc/common_audio/real_fourier_openmax.cc index e560b6cd8a..bc3e7347cb 100644 --- a/webrtc/common_audio/real_fourier_openmax.cc +++ b/webrtc/common_audio/real_fourier_openmax.cc @@ -12,7 +12,7 @@ #include -#include "third_party/openmax_dl/dl/sp/api/omxSP.h" +#include "dl/sp/api/omxSP.h" #include "webrtc/base/checks.h" namespace webrtc { diff --git a/webrtc/common_video/DEPS b/webrtc/common_video/DEPS deleted file mode 100644 index 2805958070..0000000000 --- a/webrtc/common_video/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/examples/DEPS b/webrtc/examples/DEPS deleted file mode 100644 index a562df972e..0000000000 --- a/webrtc/examples/DEPS +++ /dev/null @@ -1,7 +0,0 @@ -include_rules = [ - "+webrtc/api", - "+webrtc/base", - "+webrtc/media", - "+webrtc/modules/audio_device", - "+webrtc/p2p", -] diff --git a/webrtc/libjingle/DEPS b/webrtc/libjingle/DEPS deleted file mode 100644 index 7f492770e0..0000000000 --- a/webrtc/libjingle/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+third_party/expat", - "+webrtc/base", - "+webrtc/p2p", -] diff --git a/webrtc/libjingle/xmpp/chatroommodule_unittest.cc b/webrtc/libjingle/xmpp/chatroommodule_unittest.cc new file mode 100644 index 0000000000..65d28273cf --- /dev/null +++ b/webrtc/libjingle/xmpp/chatroommodule_unittest.cc @@ -0,0 +1,280 @@ +/* + * Copyright 2004 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include +#include +#include +#include "buzz/chatroommodule.h" +#include "buzz/constants.h" +#include "buzz/xmlelement.h" +#include "buzz/xmppengine.h" +#include "common/common.h" +#include "engine/util_unittest.h" +#include "test/unittest-inl.h" +#include "test/unittest.h" + +#define TEST_OK(x) TEST_EQ((x),XMPP_RETURN_OK) +#define TEST_BADARGUMENT(x) TEST_EQ((x),XMPP_RETURN_BADARGUMENT) + +namespace buzz { + +class MultiUserChatModuleTest; + +static void +WriteEnteredStatus(std::ostream& os, XmppChatroomEnteredStatus status) { + switch(status) { + case XMPP_CHATROOM_ENTERED_SUCCESS: + os<<"success"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_NICKNAME_CONFLICT: + os<<"failure(nickname conflict)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_REQUIRED: + os<<"failure(password required)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_INCORRECT: + os<<"failure(password incorrect)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_NOT_A_MEMBER: + os<<"failure(not a member)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_MEMBER_BANNED: + os<<"failure(member banned)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_MAX_USERS: + os<<"failure(max users)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_ROOM_LOCKED: + os<<"failure(room locked)"; + break; + case XMPP_CHATROOM_ENTERED_FAILURE_UNSPECIFIED: + os<<"failure(unspecified)"; + break; + default: + os<<"unknown"; + break; + } +} + +static void +WriteExitedStatus(std::ostream& os, XmppChatroomExitedStatus status) { + switch (status) { + case XMPP_CHATROOM_EXITED_REQUESTED: + os<<"requested"; + break; + case XMPP_CHATROOM_EXITED_BANNED: + os<<"banned"; + break; + case XMPP_CHATROOM_EXITED_KICKED: + os<<"kicked"; + break; + case XMPP_CHATROOM_EXITED_NOT_A_MEMBER: + os<<"not member"; + break; + case XMPP_CHATROOM_EXITED_SYSTEM_SHUTDOWN: + os<<"system shutdown"; + break; + case XMPP_CHATROOM_EXITED_UNSPECIFIED: + os<<"unspecified"; + break; + default: + os<<"unknown"; + break; + } +} + +//! This session handler saves all calls to a string. These are events and +//! data delivered form the engine to application code. +class XmppTestChatroomHandler : public XmppChatroomHandler { +public: + XmppTestChatroomHandler() {} + virtual ~XmppTestChatroomHandler() {} + + void ChatroomEnteredStatus(XmppChatroomModule* room, + XmppChatroomEnteredStatus status) { + RTC_UNUSED(room); + ss_ <<"[ChatroomEnteredStatus status: "; + WriteEnteredStatus(ss_, status); + ss_ <<"]"; + } + + + void ChatroomExitedStatus(XmppChatroomModule* room, + XmppChatroomExitedStatus status) { + RTC_UNUSED(room); + ss_ <<"[ChatroomExitedStatus status: "; + WriteExitedStatus(ss_, status); + ss_ <<"]"; + } + + void MemberEntered(XmppChatroomModule* room, + const XmppChatroomMember* entered_member) { + RTC_UNUSED(room); + ss_ << "[MemberEntered " << entered_member->member_jid().Str() << "]"; + } + + void MemberExited(XmppChatroomModule* room, + const XmppChatroomMember* exited_member) { + RTC_UNUSED(room); + ss_ << "[MemberExited " << exited_member->member_jid().Str() << "]"; + } + + void MemberChanged(XmppChatroomModule* room, + const XmppChatroomMember* changed_member) { + RTC_UNUSED(room); + ss_ << "[MemberChanged " << changed_member->member_jid().Str() << "]"; + } + + virtual void MessageReceived(XmppChatroomModule* room, const XmlElement& message) { + RTC_UNUSED2(room, message); + } + + + std::string Str() { + return ss_.str(); + } + + std::string StrClear() { + std::string result = ss_.str(); + ss_.str(""); + return result; + } + +private: + std::stringstream ss_; +}; + +//! This is the class that holds all of the unit test code for the +//! roster module +class XmppChatroomModuleTest : public UnitTest { +public: + XmppChatroomModuleTest() {} + + void TestEnterExitChatroom() { + std::stringstream dump; + + // Configure the engine + rtc::scoped_ptr engine(XmppEngine::Create()); + XmppTestHandler handler(engine.get()); + + // Configure the module and handler + rtc::scoped_ptr chatroom(XmppChatroomModule::Create()); + + // Configure the module handler + chatroom->RegisterEngine(engine.get()); + + // Set up callbacks + engine->SetOutputHandler(&handler); + engine->AddStanzaHandler(&handler); + engine->SetSessionHandler(&handler); + + // Set up minimal login info + engine->SetUser(Jid("david@my-server")); + engine->SetPassword("david"); + + // Do the whole login handshake + RunLogin(this, engine.get(), &handler); + TEST_EQ("", handler.OutputActivity()); + + // Get the chatroom and set the handler + XmppTestChatroomHandler chatroom_handler; + chatroom->set_chatroom_handler(static_cast(&chatroom_handler)); + + // try to enter the chatroom + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM); + chatroom->set_nickname("thirdwitch"); + chatroom->set_chatroom_jid(Jid("darkcave@my-server")); + chatroom->RequestEnterChatroom("", XMPP_CONNECTION_STATUS_UNKNOWN, "en"); + TEST_EQ(chatroom_handler.StrClear(), ""); + TEST_EQ(handler.OutputActivity(), + "" + "" + ""); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); + + // simulate the server and test the client + std::string input; + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), ""); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); + + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), ""); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER); + + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), + "[ChatroomEnteredStatus status: success]"); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM); + + // simulate somebody else entering the room after we entered + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), "[MemberEntered darkcave@my-server/fourthwitch]"); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM); + + // simulate somebody else leaving the room after we entered + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), "[MemberExited darkcave@my-server/secondwitch]"); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM); + + // try to leave the room + chatroom->RequestExitChatroom(); + TEST_EQ(chatroom_handler.StrClear(), ""); + TEST_EQ(handler.OutputActivity(), + ""); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_EXIT); + + // simulate the server and test the client + input = "" + "" + "" + "" + ""; + TEST_OK(engine->HandleInput(input.c_str(), input.length())); + TEST_EQ(chatroom_handler.StrClear(), + "[ChatroomExitedStatus status: requested]"); + TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM); + } + +}; + +// A global function that creates the test suite for this set of tests. +TestBase* ChatroomModuleTest_Create() { + TestSuite* suite = new TestSuite("ChatroomModuleTest"); + ADD_TEST(suite, XmppChatroomModuleTest, TestEnterExitChatroom); + return suite; +} + +} diff --git a/webrtc/media/DEPS b/webrtc/media/DEPS deleted file mode 100644 index 10f88d0990..0000000000 --- a/webrtc/media/DEPS +++ /dev/null @@ -1,22 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_video", - "+webrtc/modules/audio_coding", - "+webrtc/modules/audio_device", - "+webrtc/modules/audio_processing", - "+webrtc/modules/video_capture", - "+webrtc/modules/video_coding", - "+webrtc/p2p", - "+webrtc/pc", - "+webrtc/sound", - "+webrtc/system_wrappers", - "+webrtc/voice_engine", - "+usrsctplib", -] - -specific_include_rules = { - "win32devicemanager\.cc": [ - "+third_party/logitech/files/logitechquickcam.h", - ], -} diff --git a/webrtc/modules/audio_coding/DEPS b/webrtc/modules/audio_coding/DEPS deleted file mode 100644 index 31aa1c25fb..0000000000 --- a/webrtc/modules/audio_coding/DEPS +++ /dev/null @@ -1,7 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_audio", - "+webrtc/audio_coding/neteq/neteq_unittest.pb.h", # Different path. - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/audio_conference_mixer/DEPS b/webrtc/modules/audio_conference_mixer/DEPS deleted file mode 100644 index 2805958070..0000000000 --- a/webrtc/modules/audio_conference_mixer/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/audio_device/DEPS b/webrtc/modules/audio_device/DEPS deleted file mode 100644 index 2f4a597051..0000000000 --- a/webrtc/modules/audio_device/DEPS +++ /dev/null @@ -1,11 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_audio", - "+webrtc/system_wrappers", -] - -specific_include_rules = { - "ensure_initialized\.cc": [ - "+base/android", - ], -} diff --git a/webrtc/modules/audio_processing/DEPS b/webrtc/modules/audio_processing/DEPS deleted file mode 100644 index e9ac967c58..0000000000 --- a/webrtc/modules/audio_processing/DEPS +++ /dev/null @@ -1,14 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_audio", - "+webrtc/system_wrappers", -] - -specific_include_rules = { - ".*test\.cc": [ - "+webrtc/tools", - # Android platform build has different paths. - "+gtest", - "+external/webrtc", - ], -} diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc index 8c6278f40e..25b99d8773 100644 --- a/webrtc/modules/audio_processing/agc/agc_unittest.cc +++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc @@ -10,8 +10,8 @@ #include "webrtc/modules/audio_processing/agc/agc.h" -#include "testing/gmock/include/gmock/gmock.h" -#include "testing/gtest/include/gtest/gtest.h" +#include "gmock/gmock.h" +#include "gtest/gtest.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h index 9e8f64e838..e362200d86 100644 --- a/webrtc/modules/audio_processing/agc/mock_agc.h +++ b/webrtc/modules/audio_processing/agc/mock_agc.h @@ -13,7 +13,7 @@ #include "webrtc/modules/audio_processing/agc/agc.h" -#include "testing/gmock/include/gmock/gmock.h" +#include "gmock/gmock.h" #include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/bitrate_controller/DEPS b/webrtc/modules/bitrate_controller/DEPS deleted file mode 100644 index 9a462b6fc5..0000000000 --- a/webrtc/modules/bitrate_controller/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/congestion_controller/DEPS b/webrtc/modules/congestion_controller/DEPS deleted file mode 100644 index d72e34db6e..0000000000 --- a/webrtc/modules/congestion_controller/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", - "+webrtc/video", -] diff --git a/webrtc/modules/desktop_capture/DEPS b/webrtc/modules/desktop_capture/DEPS deleted file mode 100644 index 2805958070..0000000000 --- a/webrtc/modules/desktop_capture/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/include/DEPS b/webrtc/modules/include/DEPS deleted file mode 100644 index aad6d8a855..0000000000 --- a/webrtc/modules/include/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_video", -] diff --git a/webrtc/modules/media_file/DEPS b/webrtc/modules/media_file/DEPS deleted file mode 100644 index 5c5452a0c0..0000000000 --- a/webrtc/modules/media_file/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_audio", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/pacing/DEPS b/webrtc/modules/pacing/DEPS deleted file mode 100644 index 2805958070..0000000000 --- a/webrtc/modules/pacing/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/remote_bitrate_estimator/DEPS b/webrtc/modules/remote_bitrate_estimator/DEPS deleted file mode 100644 index 9a863d94a9..0000000000 --- a/webrtc/modules/remote_bitrate_estimator/DEPS +++ /dev/null @@ -1,10 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/system_wrappers", -] - -specific_include_rules = { - "nada\.h": [ - "+webrtc/voice_engine", - ], -} diff --git a/webrtc/modules/rtp_rtcp/DEPS b/webrtc/modules/rtp_rtcp/DEPS deleted file mode 100644 index 0720a15fec..0000000000 --- a/webrtc/modules/rtp_rtcp/DEPS +++ /dev/null @@ -1,6 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_video", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/utility/DEPS b/webrtc/modules/utility/DEPS deleted file mode 100644 index 1a2885bbd3..0000000000 --- a/webrtc/modules/utility/DEPS +++ /dev/null @@ -1,6 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_audio", - "+webrtc/common_video", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/video_capture/DEPS b/webrtc/modules/video_capture/DEPS deleted file mode 100644 index 58ae9fe714..0000000000 --- a/webrtc/modules/video_capture/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_video", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/video_coding/DEPS b/webrtc/modules/video_coding/DEPS deleted file mode 100644 index 512a0d8277..0000000000 --- a/webrtc/modules/video_coding/DEPS +++ /dev/null @@ -1,9 +0,0 @@ -include_rules = [ - "+third_party/ffmpeg", - "+third_party/openh264", - "+vpx", - "+webrtc/base", - "+webrtc/common_video", - "+webrtc/system_wrappers", - "+webrtc/tools", -] diff --git a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc index 7be4eb1bb2..3117e49788 100644 --- a/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc +++ b/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc @@ -11,7 +11,7 @@ #include #include -#include "testing/gtest/include/gtest/gtest.h" +#include "gtest/gtest.h" #include "vpx/vpx_encoder.h" #include "vpx/vp8cx.h" #include "webrtc/modules/video_coding/include/video_codec_interface.h" diff --git a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h index 2b2aa5de69..9f0dc5ce93 100644 --- a/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h +++ b/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h @@ -15,7 +15,6 @@ #include #include -#include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/checks.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h" @@ -23,6 +22,8 @@ #include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h" #include "webrtc/video_frame.h" +#include "gtest/gtest.h" + using ::testing::_; using ::testing::AllOf; using ::testing::Field; diff --git a/webrtc/modules/video_processing/DEPS b/webrtc/modules/video_processing/DEPS deleted file mode 100644 index 1a2885bbd3..0000000000 --- a/webrtc/modules/video_processing/DEPS +++ /dev/null @@ -1,6 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_audio", - "+webrtc/common_video", - "+webrtc/system_wrappers", -] diff --git a/webrtc/modules/video_render/DEPS b/webrtc/modules/video_render/DEPS deleted file mode 100644 index 58ae9fe714..0000000000 --- a/webrtc/modules/video_render/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_video", - "+webrtc/system_wrappers", -] diff --git a/webrtc/p2p/DEPS b/webrtc/p2p/DEPS deleted file mode 100644 index 161835f343..0000000000 --- a/webrtc/p2p/DEPS +++ /dev/null @@ -1,5 +0,0 @@ -include_rules = [ - "+net", - "+webrtc/base", - "+webrtc/system_wrappers", -] diff --git a/webrtc/pc/DEPS b/webrtc/pc/DEPS deleted file mode 100644 index ca4f789db9..0000000000 --- a/webrtc/pc/DEPS +++ /dev/null @@ -1,13 +0,0 @@ -include_rules = [ - "+webrtc/api", - "+webrtc/base", - "+webrtc/media", - "+webrtc/p2p", - "+third_party/libsrtp" -] - -specific_include_rules = { - "srtpfilter_unittest\.cc": [ - "+crypto", - ], -} diff --git a/webrtc/sound/DEPS b/webrtc/sound/DEPS deleted file mode 100644 index 7452a9fb16..0000000000 --- a/webrtc/sound/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", -] - diff --git a/webrtc/system_wrappers/DEPS b/webrtc/system_wrappers/DEPS deleted file mode 100644 index 7452a9fb16..0000000000 --- a/webrtc/system_wrappers/DEPS +++ /dev/null @@ -1,4 +0,0 @@ -include_rules = [ - "+webrtc/base", -] - diff --git a/webrtc/test/DEPS b/webrtc/test/DEPS deleted file mode 100644 index 27c0e744fd..0000000000 --- a/webrtc/test/DEPS +++ /dev/null @@ -1,13 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_video", - "+webrtc/modules/audio_coding", - "+webrtc/modules/audio_device", - "+webrtc/modules/media_file", - "+webrtc/modules/rtp_rtcp", - "+webrtc/modules/video_capture", - "+webrtc/modules/video_coding", - "+webrtc/system_wrappers", - "+webrtc/voice_engine", -] diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS deleted file mode 100644 index 73073f02f4..0000000000 --- a/webrtc/tools/DEPS +++ /dev/null @@ -1,8 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/common_video", - "+webrtc/modules/audio_processing", - "+webrtc/system_wrappers", - "+webrtc/voice_engine", -] - diff --git a/webrtc/video/DEPS b/webrtc/video/DEPS deleted file mode 100644 index 426f47c423..0000000000 --- a/webrtc/video/DEPS +++ /dev/null @@ -1,17 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_video", - "+webrtc/modules/bitrate_controller", - "+webrtc/modules/congestion_controller", - "+webrtc/modules/pacing", - "+webrtc/modules/remote_bitrate_estimator", - "+webrtc/modules/rtp_rtcp", - "+webrtc/modules/utility", - "+webrtc/modules/video_coding", - "+webrtc/modules/video_capture", - "+webrtc/modules/video_processing", - "+webrtc/modules/video_render", - "+webrtc/system_wrappers", - "+webrtc/voice_engine", -] diff --git a/webrtc/voice_engine/DEPS b/webrtc/voice_engine/DEPS deleted file mode 100644 index 224eeee676..0000000000 --- a/webrtc/voice_engine/DEPS +++ /dev/null @@ -1,14 +0,0 @@ -include_rules = [ - "+webrtc/base", - "+webrtc/call", - "+webrtc/common_audio", - "+webrtc/modules/audio_coding", - "+webrtc/modules/audio_conference_mixer", - "+webrtc/modules/audio_device", - "+webrtc/modules/audio_processing", - "+webrtc/modules/media_file", - "+webrtc/modules/pacing", - "+webrtc/modules/rtp_rtcp", - "+webrtc/modules/utility", - "+webrtc/system_wrappers", -]